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)
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.
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
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
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.
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
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
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
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
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
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
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
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
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
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