Dropped internal call from polycom but not softphone

Im actually using Allstar (www.allstarlink.org), but since its guts are asterisk I figured i’d post here first. For some background, allstar is special packaging of asterisk with a module (apt_rpt) that is use for controlling amateur radio repeaters. You can find out more at link above if interested.

I’ve got the radio side of things working well, but im having an issue on the phone side of things. I’ve set up 3 extensions. One is a Polycom Soundpoint 550, and the other two are Zoiper running on android devices. All 3 devices are on my internal 192.168.1.0/24 subnet. All 3 extensions can call out via my sip provider with out problem… no disconnects, no dropped calls. Works perfect. The problem lies when i connect to one of my “radio nodes”. The calling part of it works fine, but one of the things apt_rpt does is play messages, like ID’s and such. When something like that plays i see something like this in the asterisk console:

-- <DAHDI/pseudo-672364238> Playing 'rpt/node' (language 'en')
-- <DAHDI/pseudo-672364238> Playing 'digits/4' (language 'en')
-- <DAHDI/pseudo-672364238> Playing 'digits/2' (language 'en')
-- <DAHDI/pseudo-672364238> Playing 'digits/1' (language 'en')
-- <DAHDI/pseudo-672364238> Playing 'digits/9' (language 'en')
-- <DAHDI/pseudo-672364238> Playing 'digits/0' (language 'en')
-- <DAHDI/pseudo-672364238> Playing 'rpt/repeat_only' (language 'en')
-- Hungup 'DAHDI/pseudo-672364238'

In this case I hit *70 on the radio DTMF pad which tells it to give me the node status. It simply plays some wav’s to say the node number (42190) and “repeat only” since no other nodes are connected.Then that channel (I think thats the right term) “hangs up” . Well 32ish seconds after that hangup, my sip phone hangs up. It seems like the phone thinks the call ended and I speculate that it has something to do with:

– Hungup ‘DAHDI/pseudo-672364238’

thats causing it but im not super versed in the asterisk world yet (but im getting there i think), so i have no idea where to look. I initially thought it was a timeout from all the forum posts ive read all over the internet, but im not quite so sure thats the case. A couple notes:

Both of the softphones do not experience this issue
All 3 “phones” can call each other and out via sip truck without this issue.

snip from sip.conf

[5001]
deny=0.0.0.0/0.0.0.0
username=5001
secret=XXXXXX
dtmfmode=rfc2833
canreinvite=no
context=mobileuser
host=dynamic
trustrpid=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
sendrpid=no
type=friend
nat=auto
qualify=yes
port=5060
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/5001
mailbox=
permit=0.0.0.0/0.0.0.0
callerid=ALLSTAR <000005001>

Any troubleshooting advice would be appreciated!

Something i forgot to add. I do think this is phone related or some setting in the phone im missing, since it does not happen on the softphones. Im looking into getting another brand sip phone to test with, but im still interested to hear if anyone has any feedback or troubleshooting advice.