Re: [asterisk-users] Still sipping frustration - only getting state ACK
On Sat, 2010-06-05 at 22:16 +0200, Julien Claassen wrote: But when I make a call; channel originate sip/iptel-out/e...@iptel.org Application playback vm/net_ring The call is onlyleft in state ACK for a while. Then asterisk tells me, that it is destroying the sip dialog (long ID) INVITE. This could be caused by a number of reasons, but the most likely is that your syntax isn't correct above. Try either: channel originate sip/iptel-out/echo Application playback vm/net_ring or channel originate sip/e...@iptel-out Application playback vm/net_ring -- 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] Still sipping frustration - only getting state ACK
Hello Jared! OK, now calls go in and out. Even with the syntax: channel originate sip/mu...@iptel.org application ... it works. I've tested that with application record. But, the channel only displays ACK and core show channels doesn't list it as a call or a processed call afterwards. But thanks for the tipp, that got me thinking. Kindly yours Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] Still sipping frustration - only getting state ACK
Hello everyone! So now I found someone to forward the ports 5060 and 16000-16100 on my router and made sure to enter these ports 16000-16100 in rtp.conf. Still I get no calls going. The call is initiated. sip show channels shows the call with status ACK and then the dialog with method invite is destroyed.I tried both using application jack and application playback. So what else can be the problem? Kindly yours Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] Still sipping frustration - only getting state ACK
At 08:43 AM 6/6/2010, you wrote: So now I found someone to forward the ports 5060 and 16000-16100 on my router and made sure to enter these ports 16000-16100 in rtp.conf. Still I get no calls going. I should point out, that I just realized I've not a clue what app jack is. I use sip and forwarding those ports is necessary for sip. Have you tried with just a SIP soft phone? 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] Still sipping frustration - only getting state ACK
Hi Ira! Sorry, can't use any softphone, to my knowledge. They all come with GUIs or don't support JACK or have so limited ALSA support, that they don't fit my card (which has a lot of channels and some other HD-recording stuff). Still I did try the sip call with app playback as well. That's a very simple one, which has proved reliable countless times before. The file I use to playback is a simple mono 8kHz 16bit .WAV file, stored in the correct place (/var/lib/asterisk/sounds). So where to go now? Is there a test - without asterisk -, that I can perform to double check that the ports are correctly forwarded? Or would this be pointless, seeing that the registration works fine? Kindly yours Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] Still sipping frustration - only getting state ACK
Hello all! Hm, I just examined the output of chan_sip's debug again and found this, might that be the problem: Warning: 392 213.192.59.75:5060 Noisy feedback tells: pid=3955 req_src_ip=91.58.9.172 req_src_port=24002 in_uri=sip:sip.iptel.org out_uri=sip:sip.iptel.org via_cnt==1 I don't have configured port 24002, or is this on iptel's site? If not, how can I change it. I suppose changing this is possible. and what is that about noisy feedback? Is it a problem, or can it be ignored for the time being? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] Still sipping frustration - only getting state ACK
At 11:08 AM 6/6/2010, you wrote: So where to go now? Is there a test - without asterisk -, that I can perform to double check that the ports are correctly forwarded? Or would this be pointless, seeing that the registration works fine? I wish I could help. My one and only Linux experience is with Asterisk. I have a Digium 4 port analog board for POTS calls and the rest is SIP including all the phones in the house. Once I built the current tom based machine, and upgraded to 1.6 it's been rock solid. 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] Still sipping frustration - only getting state ACK
Thanks anyway, Ira. It was very kind of you to help me along as far as you could. I appreciate it. anyone else here, who might be able to help me along with my problem? Warmly yours Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] Still sipping frustration - only getting state ACK
Julien, Just for the record, you don't need registration to iptel.org - just plain DIAL(SIP/iptel/music). On Sun, Jun 6, 2010 at 11:47 PM, Julien Claassen jul...@c-lab.de wrote: Thanks anyway, Ira. It was very kind of you to help me along as far as you could. I appreciate it. anyone else here, who might be able to help me along with my problem? Warmly yours Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] Still sipping frustration - only getting state ACK
Julien Claassen wrote: Hello everyone! I still am not much further along with my sip calling. I changed my sip.conf taking suggestions from the net (voip-info.org in particular). I changed iptel's position from friend to peer. I turned on and off nat, I chose different codecs in first place, entered my outward IP as fromdomain and uncommented the register directive with correct values. All I get is two registrations now, but no calls. get a registration effort every 225secs and it succeeds. But when I make a call; channel originate sip/iptel-out/e...@iptel.org Application playback vm/net_ring The call is onlyleft in state ACK for a while. Then asterisk tells me, that it is destroying the sip dialog (long ID) INVITE. Question: Might it be a problem, that my system only knows itself as 192.168.*. Do I need to set something else than externip? Does the server see your sip client at 192.168.*.*? that would be a problem. Might it be, that my router really blocks certain ports? I can't check it, since it's heavily javascript based and, since I'm blind and the accessibility software for the GUI never really worked on this distro, I don't have a browser to look at it. It's possible that the router is not SIP friendly or there is a setting to allow sip on it. I can not tell as I don't know what router you are using. Do I need to forward port 5060 to my machine specifically (like it is needed for SSH's port 22), or is the mechanism based on: I talk first and the sever gets back to me based on that. Should not need any forwards. However the router could be firewalling some ports, like the rtp ports. You need to ask what ports are needed for rtp. Lyle Giese LCR Computer Services, Inc. This configuration worked for googletalk. I admit, there were problems, but calls were coming through from both sides. Please can someone help me clear up this mess. I'm completely frustrated and don't know what else to do, where else to look. Kindly yours and thanks in advance JUlien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] Still sipping frustration - only getting state ACK
At 01:16 PM 6/5/2010, you wrote: Please can someone help me clear up this mess. I'm completely frustrated and don't know what else to do, where else to look. I've always forwarded port 5060 and all the RTP ports, in my case 16000-16100, directly to my Asterisk box and I've never had problems. For SIP you really need to forward the RTP ports you expect to use and you might as well forward 5060 as that's the only place you want it to go. Or, that's my belief. Try it and see, but find out what RTP ports you need by looking in rtp.conf for the lines: rtpstart=16000 rtpend=16100 I picked those numbers because I thought using the default huge range made no sense and I've never had a problem with my no more than 3 calls at a time world. 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] Still sipping frustration - only getting state ACK
Hello Lyle! Thanks for your answer! I don't know, if the server sees me at the local-ip or not. I only know, that I'm able to register at iptel.org successfully. So asterisk tells me. I believe my router is a Samsung router 3010 phone SL. Samsung it tells me, the rest I had to search on the web, I'm with freenet for a while. How can I find out, which IP the server sees and knows of me? I know that in some of the debug/notice messages from chan_sip I saw aster...@127.0.0.1. Kindest regards and thanks again Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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] Still sipping frustration - only getting state ACK
Hello Ira! I will have a look at my rtp.conf and change the rtp-port range there. As to forwarding: Well it remains to be seen - pardon the pun - if I can find someone willing and patient enough to be my pair of eyes. :-) Kindly yours Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de -- _ -- 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
