Can one of you developer please explain what is going on here and what these error messages mean?
It seems everyone who gets these messages is having the same problem in that they can’t hear voices from there calls, which appears to me because asterisk is dropping the bridge between the SIP Provider, and the Client…
Didn’t get a frame from channel
Bridge stops bridging channels
Thanks,
Daniel A. Creed
$50 offer to anyone who can resolve this problem for me…
Thanks,
Dan
Do the developers read these forums?
no, developers don’t read these forums. they are staffed by volunteers, mostly asterisk users. There is probably not much ‘helper’ going on now because of the holiday.
didnt get a frame from chanel i believe means * was looking for a voice frame and didn’t get one. bridge stops bridging means your calls audio path is being disconnected.
sounds like you could have NAT issues…
if you want a more helpful response please post more info, see the sticky at the top of the forum forums.digium.com/viewtopic.php?t=4208 for what is good to post…
NO NAT… I’ve got my asterisk box configured with a public IP address… my client to my asterisk box is of course NAT’ed 192.168.10.X…
I use broad voice… here is my sip_additional.conf (using freepbx)
register=2627350025@sip.broadvoice.com:9639wx29q9:2627350025@sip.broadvoice.com
[210]
username=210
type=friend
secret=1234
record_out=Always
record_in=Always
qualify=yes
port=5060
nat=yes
mailbox=210@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=yes
callerid=House Phone <210>
[broadvoice-in]
username=2627350025
user=2627350025
type=user
secret=9639wx29q9
nat=no
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-pstn
[broadvoice-out]
username=2627350025
user=phone
type=peer
secret=9639wx29q9
nat=no
insecure=very
host=sip.broadvoice.com
fromuser=2627350025
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-pstn
canreinvite=no
authname=2627350025
Here is the log from a call
Dec 22 11:57:46 VERBOSE[3412] logger.c: – SIP/broadvoice-out-0a1b11f0 is making progress passing it to SIP/201-0a1b61d8
Dec 22 11:57:46 DEBUG[3412] channel.c: Building translator from ulaw to SLINEAR for spies on channel SIP/201-0a1b61d8
Dec 22 11:57:52 DEBUG[3316] chan_sip.c: Acked pending invite 103
Dec 22 11:57:52 DEBUG[3316] chan_sip.c: Stopping retransmission on '3d2a445b5df20cfb0a46aee13a67a663@sip.broadvoice.com’ of Request 103: Match Found
Dec 22 11:57:52 DEBUG[3316] chan_sip.c: build_route: Contact hop: sip:4142326957@147.135.12.128
Dec 22 11:57:52 VERBOSE[3412] logger.c: – SIP/broadvoice-out-0a1b11f0 answered SIP/201-0a1b61d8
Dec 22 11:57:52 DEBUG[3316] chan_sip.c: Stopping retransmission on ‘b4bc2c49-90302803@192.168.10.224’ of Response 102: Match Found
Dec 22 11:57:55 DEBUG[3316] chan_sip.c: Scheduled a registration timeout for sip.broadvoice.com id #134
Dec 22 11:57:56 DEBUG[3316] chan_sip.c: Stopping retransmission on ‘6fd23abf0bd9ca823cf2e6d64e9e9dcd@127.0.0.1’ of Request 115: Match Found
Dec 22 11:57:56 DEBUG[3316] chan_sip.c: Registration successful
Dec 22 11:57:56 DEBUG[3316] chan_sip.c: Cancelling timeout 134
Dec 22 11:58:03 DEBUG[3316] chan_sip.c: Auto destroying call '6fd23abf0bd9ca823cf2e6d64e9e9dcd@127.0.0.1’
Dec 22 11:58:07 DEBUG[3316] chan_sip.c: Stopping retransmission on ‘1bb890ea3462cad433ed406f3950b0ef@192.168.10.16’ of Request 102: Match Found
Dec 22 11:58:07 DEBUG[3316] chan_sip.c: Stopping retransmission on ‘2d712143695a5a3815a258f109c1161e@192.168.10.16’ of Request 102: Match Found
Dec 22 11:58:07 DEBUG[3316] chan_sip.c: Stopping retransmission on ‘2d6990ea1155e0cd6e73a99f0f35073e@192.168.10.16’ of Request 102: Match Found
Dec 22 11:58:07 DEBUG[3412] channel.c: Didn’t get a frame from channel: SIP/201-0a1b61d8
Dec 22 11:58:07 DEBUG[3412] channel.c: Bridge stops bridging channels SIP/201-0a1b61d8 and SIP/broadvoice-out-0a1b11f0
Dec 22 11:58:07 DEBUG[3412] chan_sip.c: update_call_counter(4142326957) - decrement call limit counter
Dec 22 11:58:07 DEBUG[3412] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Dec 22 11:58:07 VERBOSE[3412] logger.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on ‘SIP/201-0a1b61d8’ in macro ‘dialout-trunk’
Dec 22 11:58:07 VERBOSE[3412] logger.c: == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on ‘SIP/201-0a1b61d8’
Dec 22 11:58:07 VERBOSE[3412] logger.c: – Executing Macro(“SIP/201-0a1b61d8”, “hangupcall”) in new stack
Dec 22 11:58:07 VERBOSE[3412] logger.c: – Executing ResetCDR(“SIP/201-0a1b61d8”, “w”) in new stack
Dec 22 11:58:07 DEBUG[3412] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Dec 22 11:58:07 DEBUG[3412] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES (‘2006-12-22 11:57:43’,‘2627350025’,‘2627350025’,‘4142326957’,‘from-internal’, ‘SIP/201-0a1b61d8’,‘SIP/broadvoice-out-0a1b11f0’,‘ResetCDR’,‘w’,24,15,‘ANSWERED’,3,’’)
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '2627350025’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '2627350025’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '4142326957’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is 'from-internal’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is 'SIP/201-0a1b61d8’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is 'SIP/broadvoice-out-0a1b11f0’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is 'ResetCDR’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is 'w’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '2006-12-22 11:57:43’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '2006-12-22 11:57:52’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '2006-12-22 11:58:07’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '24’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '15’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is 'ANSWERED’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is 'DOCUMENTATION’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '(null)'
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '1166810263.6’
Dec 22 11:58:07 DEBUG[3412] pbx.c: Function result is '(null)'
Dec 22 11:58:07 VERBOSE[3412] logger.c: – Executing NoCDR(“SIP/201-0a1b61d8”, “”) in new stack
Dec 22 11:58:07 NOTICE[3412] cdr.c: CDR on channel ‘SIP/201-0a1b61d8’ not posted
Dec 22 11:58:07 NOTICE[3412] cdr.c: CDR on channel ‘SIP/201-0a1b61d8’ lacks end
Dec 22 11:58:07 DEBUG[3412] pbx.c: Expression result is '1’
Dec 22 11:58:07 VERBOSE[3412] logger.c: – Executing GotoIf(“SIP/201-0a1b61d8”, “1?theend”) in new stack
Dec 22 11:58:07 VERBOSE[3412] logger.c: – Goto (macro-hangupcall,s,6)
Dec 22 11:58:07 VERBOSE[3412] logger.c: – Executing Wait(“SIP/201-0a1b61d8”, “5”) in new stack
Dec 22 11:58:07 VERBOSE[3412] logger.c: == Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/201-0a1b61d8’ in macro 'hangupcall’
Dec 22 11:58:07 VERBOSE[3412] logger.c: == Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/201-0a1b61d8’
Dec 22 11:58:07 DEBUG[3412] channel.c: Spy MixMonitor removed from channel SIP/201-0a1b61d8
Dec 22 11:58:07 DEBUG[3412] chan_sip.c: update_call_counter(201) - decrement call limit counter
ah, i think your client needs to be set up with STUN, or told what its public IP is. THen it will report the correct IP to asterisk (this is NOT done with the packet’s FROM header) and it will work correctly. Right now Asterisk is looking for packets from 192.168.whatever and not getting them, thus your error.
If you tell the client what its public IP is or set it up with STUN (make it figure this out for itself) then it will report the right IP and will work.
I will try this at home tonight… but I’m not sure how to setup a STUN server… will need to look into that more… plus I’m not 100% sure that is the problem, because this setup WORKED for months… but then just quit working after I upgraded to the latest release…
Plus I am having the same problem, where now I am using a client and the server with NO NAT (both have public IP) and I am still having the same problem…
I can hear the caller… but the caller can’t hear me…
Thanks,
Dan
hmmm that is strange. as for STUN you dont need to set up your own use stun.xten.net or stun.softjoys.com as i recall. stun just helps a client figure out what kind of NAT its behind.
i assume you have externip= and localnet= set correctly in sip.conf?
also try setting canreinvite=no everywhere…
I’ve tried setting a sip_nat.conf with externip etc… but according to all the posts and the documents you don’t need that when Asterisk isn’t NAT’ed… which in my case its not… I’ve tried it anyways… still no luck… I’ve also tried setting canreinvite to no everywhere… still no luck…
Thanks,
Dan
Wow… no one has taken me up on this even though I am offering $50 for a solution… Somebody out there has to be able to help me out…
-Dan