Thanks for your help. This is all running on amazon AWS so perhaps I do have NAT issues. However, if that's the case I don't get why media flows when I don't add 'dtmf_events' to the bridge type, and why it works when I do have that bridge type with both sides using the working trunk. Is it because asterisk detects different dtmf modes between the two outside endpoints and only then tries to have media flow through itself on our network?
Here's my pjsip.conf:
[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0
[workingdtmftrunk]
type=auth
auth_type=userpass
password=<password>
username=<username>
realm=sip.flowroute.com
[workingdtmftrunk]
type=aor
contact=sip:sip.flowroute.com
[workingdtmftrunk]
type=registration
transport=simpletrans
outbound_auth=workingdtmftrunk
server_uri=sip:sip.flowroute.com
client_uri=sip:<username>@sip.flowroute.com
contact_user=<user>
retry_interval=60
[workingdtmftrunk]
type=endpoint
context=inbound
disallow=all
allow=ulaw
allow=g729
transport=simpletrans
outbound_auth=workingdtmftrunk
aors=workingdtmftrunk
[workingdtmftrunk]
type=identify
endpoint=workingdtmftrunk
match=sip.flowroute.com
[baddtmftrunk]
type=aor
contact=sip:<ip>
[baddtmftrunk]
type=endpoint
disallow=all
allow=ulaw
transport=simpletrans
aors=baddtmftrunk
dtmf_mode=info
On Thu, Oct 16, 2014 at 10:27 AM, Joshua Colp <jcolp@xxxxxxxxxx> wrote:
Nick Horelik wrote:
Hello,
Kia ora,
I've been using ARI for a few months to build some basic apps with some
success - I definitely like the interface!
Glad to hear it! If there's any improvements you can think of don't hesitate to make them known. It's important to get feedback from people actually using it.
Lately I've run into an issue that I'm not sure how to solve, and I
wanted to ask about best ways to debug. I'm still a little new to
Asterisk in general, so I apologize in advance if I'm missing something
simple.
Right now I'm trying to build a simple app to test the registration of
DTMF events over a specific trunk that doesn't use rfc4733 for dtmf_mode
- they only support rfc2833 and sip-info. I figured out by looking at
the source of res/res_pjsip/pjsip_configuration.c that I can set
dmtf_mode=info on my endpoint in pjsip.conf (I had trouble finding
documentation on this - is there somewhere I can look for more info
about dtmf_mode?), so I'm hoping that this will work for me. I didn't
see the option to use rfc2833 with pjsip.
The rfc4733 option is rfc2833 pretty much, you can use that without any problem.
<snip>
Everything works fine if I send the 'outgoing' channel over
<working_dtmf_trunk>: the cellphones are connected successfully (I can
hear audio from one to the other) and my function registered to
ChannelDtmfReceived successfully gets triggered when I press keys on
cellphone_2. The cellphones are also successfully connected when using
<bad_dtmf_trunk> for the 'outgoing' channel, when I use a bridge created
with:
client.bridges.create(type='mixing')
Of course in this case the dtmf events are not captured. Here's where
I'm stuck: when I use 'mixing,dtmf_events' for the bridge to connect the
'outgoing' channel over <bad_dtmf_trunk>, the phones do not seem to be
successfully connected. The second cellphone rings and I can answer the
call, but I can't hear audio between the phones and my function
registered to ChannelDtmfReceived is never triggered.
Am I missing something fundamental with what I'm trying to do here? I'm
happy to post my pjsip.conf if you think my issue might lie there. If
everything looks good I can try working with support at this specific
trunk, but I want to make sure the issue isn't on my end before pursuing
that.
I don't think you are having an ARI problem, it sounds like you are having a NAT issue. Are you behind one by chance? For reliable media flow there's extra configuration and port forwarding that should occur. If it doesn't then you'll get exactly that behavior. In the working case Asterisk will, by default, have media flow directly between both sides which if they are both public and outside your network will have it work. Your pjsip.conf would also provide a better glimpse into the exact setup.
Separately from the specifics of this example, what's the right way to
debug these sorts of issues? I've tried checking
/var/log/asterisk/messages, setting up logs with cel_custom.conf,
watching the CLI while I try to test - I haven't yet found the right way
to get any more visibility into why it's not working. At this point I'm
considering capturing packets to investigate it, but I feel like there
must be a better way. Also, is there an easy way to dump to a log all
events that flow through an ARI application, and not just ones I've
registered functions to in the app itself?
A packet capture and looking at the SIP signaling + SDP is the easiest way for no media flow scenarios.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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