Re: [asterisk-users] RTP not being switched between both SIP endpoints
Hi, Since opensips is not handling media (i presume) is the audio not already going direct from asterisk to the other endpoint? Thanks Kenny From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Gareth Blades [mailinglist+aster...@dns99.co.uk] Sent: 17 September 2013 11:17 To: asterisk-users@lists.digium.com Subject: [asterisk-users] RTP not being switched between both SIP endpoints We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call coming from opensips. Directmedia is not specifically defined so its using the asterisk default value. * Name : vmpubopensips3 Description : Secret : Not set MD5Secret: Not set Remote Secret: Not set Context : from-pubopensips Record On feature : automon Record Off feature : automon Subscr.Cont. : Not set Language : Tonezone : Not set AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : no Force rport : Auto (No) Symmetric RTP: No ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 88.x.x.x Addr-IP : 88.x.x.x:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : (gsm|ulaw|alaw) Codec Order : (alaw:20,ulaw:20,gsm:20) Auto-Framing : No Status : Unmonitored Useragent: Reg. Contact : Qualify Freq : 6 ms Keepalive: 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No When the call comes in the SDP contains :- v=0. o=root 973184584 973184584 IN IP4 81.x.x.x s=session. c=IN IP4 81.x.x.x t=0 0. m=audio 11370 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and we reply back with :- v=0. o=root 822402971 822402971 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10428 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. When we send the outbound SIP information we advertise the following SDP :- v=0. o=root 431105643 431105643 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10144 RTP/AVP 8 3 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and the other end replies with :- v=0. o=hksbc1a 609621538 609621538 IN IP4 203.x.x.x s=sip call. c=IN IP4 203.x.x.x t=0 0. m=audio 34146 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. In the Dial() command the only option we are using is M() which is used to run a macro when the call is answered. This is used to update cdr records and perform other features if they are enabled. In this case we are just updating the cdr records so I would expect the audio to be switched as soon as the macro finishes. Any ideas what could be wrong? We are running Asterisk PBX 11.2-cert2 Thanks Gareth -- _ -- 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
Re: [asterisk-users] RTP not being switched between both SIP endpoints
On 18/09/13 12:40, Kenny Watson wrote: Hi, Since opensips is not handling media (i presume) is the audio not already going direct from asterisk to the other endpoint? Thanks Kenny Opensips wasnt handling the media so the audio was between the original caller and asterisk (with the signalling being relayed by opensips). It was just when we dialled onto the final destination via SIP asterisk stayed in the loop and didnt issue a reinvite. Its all fixed now. Although we weren't using any features the AGI application was setting DYNAMIC_FEATURES to an empty string which was enough to keep asterisk in a loop. We stopped the AGI from setting the variable if there were no features and it started working. Thanks Gareth -- _ -- 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] Can't connect to Asterisk cli
Hello, I have started using Asterisk recently on my Ubuntu server. I installed it first using apt-get and it worked fine sort of, but still couldn't hear voice during the call! I read that this problem solved by reinstalling it, so I decided to reinstall the latest version from the source as apt-get can give you only Asterisk 1.8So I have installed Asterisk 11 following the procedure in Asterisk- The Definitive Guide, 4th Edition bookThe installation done with no problem, but now I can't even login to CLI, it keeps returns Linux prompt! ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -rubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vcubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvvubuntu@vbefe01:/etc/asterisk$ and if I don't use sudo, it returns a core dump$ asterisk -vvcIllegal instruction (core dumped) I was able to connect before installing the latest version from the source? How can I fix that? Thanks. -- _ -- 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] Can't connect to Asterisk cli
become root sudo su - or su -l give your password. if asterisk is already running connect to asterisk -rvvvc otherwisw asterisk -c. if you want asterisk run as daemon asterisk and then connect to asterisk asterisk -rvvvc On Wed, Sep 18, 2013 at 2:13 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have started using Asterisk recently on my Ubuntu server. I installed it first using apt-get and it worked fine sort of, but still couldn't hear voice during the call! I read that this problem solved by reinstalling it, so I decided to reinstall the latest version from the source as apt-get can give you only Asterisk 1.8 So I have installed Asterisk 11 following the procedure in Asterisk- The Definitive Guide, 4th Edition book The installation done with no problem, but now I can't even login to CLI, it keeps returns Linux prompt! ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv ubuntu@vbefe01:/etc/asterisk$ and if I don't use sudo, it returns a core dump $ asterisk -vvc Illegal instruction (core dumped) I was able to connect before installing the latest version from the source? How can I fix that? Thanks. -- _ -- 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] Can't connect to Asterisk cli
Have you checked your SELinux settings? On 18 September 2013 13:13, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have started using Asterisk recently on my Ubuntu server. I installed it first using apt-get and it worked fine sort of, but still couldn't hear voice during the call! I read that this problem solved by reinstalling it, so I decided to reinstall the latest version from the source as apt-get can give you only Asterisk 1.8 So I have installed Asterisk 11 following the procedure in Asterisk- The Definitive Guide, 4th Edition book The installation done with no problem, but now I can't even login to CLI, it keeps returns Linux prompt! ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv ubuntu@vbefe01:/etc/asterisk$ and if I don't use sudo, it returns a core dump $ asterisk -vvc Illegal instruction (core dumped) I was able to connect before installing the latest version from the source? How can I fix that? Thanks. -- _ -- 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 -- 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] Can't connect to Asterisk cli
SELinux exists in Ubuntu? On Wed, Sep 18, 2013 at 2:45 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Have you checked your SELinux settings? On 18 September 2013 13:13, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have started using Asterisk recently on my Ubuntu server. I installed it first using apt-get and it worked fine sort of, but still couldn't hear voice during the call! I read that this problem solved by reinstalling it, so I decided to reinstall the latest version from the source as apt-get can give you only Asterisk 1.8 So I have installed Asterisk 11 following the procedure in Asterisk- The Definitive Guide, 4th Edition book The installation done with no problem, but now I can't even login to CLI, it keeps returns Linux prompt! ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv ubuntu@vbefe01:/etc/asterisk$ and if I don't use sudo, it returns a core dump $ asterisk -vvc Illegal instruction (core dumped) I was able to connect before installing the latest version from the source? How can I fix that? Thanks. -- _ -- 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 -- 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 -- _ -- 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] Can't connect to Asterisk cli
It looks this is because Asterisk isn't startedwhen tried to start it, I got a core dump!$ /etc/init.d/asterisk start * Starting Asterisk PBX: asteriskIllegal instruction (core dumped) From: asabatg...@hotmail.com To: asterisk-users@lists.digium.com Subject: Can't connect to Asterisk cli Date: Wed, 18 Sep 2013 14:13:16 +0200 Hello, I have started using Asterisk recently on my Ubuntu server. I installed it first using apt-get and it worked fine sort of, but still couldn't hear voice during the call! I read that this problem solved by reinstalling it, so I decided to reinstall the latest version from the source as apt-get can give you only Asterisk 1.8So I have installed Asterisk 11 following the procedure in Asterisk- The Definitive Guide, 4th Edition bookThe installation done with no problem, but now I can't even login to CLI, it keeps returns Linux prompt! ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -rubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vcubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvvubuntu@vbefe01:/etc/asterisk$ and if I don't use sudo, it returns a core dump$ asterisk -vvcIllegal instruction (core dumped) I was able to connect before installing the latest version from the source? How can I fix that? Thanks. -- _ -- 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] Can't connect to Asterisk cli
Ah, my bad, possibly not. On 18 September 2013 13:50, Asghar Mohammad asghar...@gmail.com wrote: SELinux exists in Ubuntu? On Wed, Sep 18, 2013 at 2:45 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Have you checked your SELinux settings? On 18 September 2013 13:13, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have started using Asterisk recently on my Ubuntu server. I installed it first using apt-get and it worked fine sort of, but still couldn't hear voice during the call! I read that this problem solved by reinstalling it, so I decided to reinstall the latest version from the source as apt-get can give you only Asterisk 1.8 So I have installed Asterisk 11 following the procedure in Asterisk- The Definitive Guide, 4th Edition book The installation done with no problem, but now I can't even login to CLI, it keeps returns Linux prompt! ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv ubuntu@vbefe01:/etc/asterisk$ and if I don't use sudo, it returns a core dump $ asterisk -vvc Illegal instruction (core dumped) I was able to connect before installing the latest version from the source? How can I fix that? Thanks. -- _ -- 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 -- 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 -- _ -- 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 -- 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] Can't connect to Asterisk cli
i think you messed 2 installs of asterisk. if you compile asterisk from sources it not insert init script. you can test installing to /opt. 1. cd to asterisk sources folder 2. make distclean 3. ./configure --prefix=/opt/asterisk 4. make 5. sudo make install 6. /opt/asterisk/sbin/asterisk -c you can remove this installation by sudo rm -rfv /opt/asterisk if it work then you should remove every asterisk installation and then install a fresh copy .(or reinstall OS if you cannot remove) hope this will help you. On Wed, Sep 18, 2013 at 2:57 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: It looks this is because Asterisk isn't started when tried to start it, I got a core dump! $ /etc/init.d/asterisk start * Starting Asterisk PBX: asterisk Illegal instruction (core dumped) -- From: asabatg...@hotmail.com To: asterisk-users@lists.digium.com Subject: Can't connect to Asterisk cli Date: Wed, 18 Sep 2013 14:13:16 +0200 Hello, I have started using Asterisk recently on my Ubuntu server. I installed it first using apt-get and it worked fine sort of, but still couldn't hear voice during the call! I read that this problem solved by reinstalling it, so I decided to reinstall the latest version from the source as apt-get can give you only Asterisk 1.8 So I have installed Asterisk 11 following the procedure in Asterisk- The Definitive Guide, 4th Edition book The installation done with no problem, but now I can't even login to CLI, it keeps returns Linux prompt! ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -r ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -vc ubuntu@vbefe01:/etc/asterisk$ sudo asterisk -cvvv ubuntu@vbefe01:/etc/asterisk$ and if I don't use sudo, it returns a core dump $ asterisk -vvc Illegal instruction (core dumped) I was able to connect before installing the latest version from the source? How can I fix that? Thanks. -- _ -- 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] sipgate outgoing calls
Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '01179553708 sip:sip...@sipgate.co.uk;tag=as30eb9dd1' -- SIP/sipgate-014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register = SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri sk any suggestions ? Many thanks -- _ -- 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] RTP port ranges
Hello Thorsten, Tuesday, September 17, 2013, 1:05:15 AM, you wrote: Where is it stated that you MUST use 1-2 ??? Someone else please ? Well, I don't use that range. This is that part of my rtp.conf rtpstart=16000 rtpend=16100 I knew I didn't need the default 25000 ports, in fact 100 is probably more than 10 times what I'll ever need. Been working for for 5 years with those numbers. I decided when I first did this that if I used non standard ports I might be less susceptible to hacking. Probably not accurate, but I did it anyway. -- Ira-- _ -- 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] sipgate outgoing calls
Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit : Hello Hi i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '01179553708 sip:sip...@sipgate.co.uk;tag=as30eb9dd1' -- SIP/sipgate-014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register = SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri sk any suggestions ? Many thanks My setup with sipgate.de [sipgate] type=peer secret=MY-PASSWORD defaultuser=SIP-ID host=217.10.79.9 fromuser=SIP-ID fromdomain=sipgate.de context=incoming-sipgate ;qualify=900 dtmfmode=info directmedia=yes insecure=port,invite disallow=all allow=ulaw,alaw accountcode=MY-ACCOUNTCODE What you forget is to register with them: ; Sipgate register = SIP-ID:my-passw...@sipgate.de/SIP-ID ;don't accept to register without FQDN Hope that help -- Daniel -- _ -- 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] sipgate outgoing calls
I believe registration is in place, otherwise inbound calls would not work. Also, registration is not required for outbound calls to work. I would suggest cutting down your sip.conf profile to this minimal configuration: host=sipgate.co.uk username=xxx fromuser=xxx insecure=invite,port secret=xxx context=my-inbound-context type=peer If outbound calls still do not with this, I would suggest that there may be an issue in the general section of your sip.conf Assuming calls do work, you can then add any other configuration lines you feel are necessary - but remember, as with all Asterisk configuration files, less is more :-) On 18 Sep 2013 22:06, Administrator TOOTAI ad...@tootai.net wrote: Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit : Hello Hi i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '01179553708 sip:sip...@sipgate.co.uk;**tag=as30eb9dd1' -- SIP/sipgate-014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register = SIP-ID:SIP-Password@sipgate.**co.uk/SIP-IDhttp://SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.**sipgate.co.ukhttp://proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/**article/394/How_do_I_**configure_Asteriskhttp://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk http://www.live.sipgate.co.uk/**faq/article/508/How_do_I_** configure_Asterihttp://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri sk any suggestions ? Many thanks My setup with sipgate.de [sipgate] type=peer secret=MY-PASSWORD defaultuser=SIP-ID host=217.10.79.9 fromuser=SIP-ID fromdomain=sipgate.de context=incoming-sipgate ;qualify=900 dtmfmode=info directmedia=yes insecure=port,invite disallow=all allow=ulaw,alaw accountcode=MY-ACCOUNTCODE What you forget is to register with them: ; Sipgate register = SIP-ID:my-passw...@sipgate.de/**SIP-IDhttp://SIP-ID:my-passw...@sipgate.de/SIP-ID;don't accept to register without FQDN Hope that help -- Daniel -- __**__**_ -- 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-**usershttp://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] RTP port ranges
I only use 100 ports as well but we have a very low call volume. I thought that I saw that you need to allocate 2 ports for every simultaneous call that you need to support. The ports are free (no charge) and are UDP not TCP so you do not lose any TCP ports. I am not sure what a hacker could do if they attacked these ports. Ron On 18/09/2013 2:29 PM, Ira wrote: Re: [asterisk-users] RTP port ranges Hello Thorsten, Tuesday, September 17, 2013, 1:05:15 AM, you wrote: Where is it stated that you MUST use 1-2 ??? Someone else please ? Well, I don't use that range. This is that part of my rtp.conf rtpstart=16000 rtpend=16100 I knew I didn't need the default 25000 ports, in fact 100 is probably more than 10 times what I'll ever need. Been working for for 5 years with those numbers. I decided when I first did this that if I used non standard ports I might be less susceptible to hacking. Probably not accurate, but I did it anyway. -- Ira -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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
