Gizmo incoming peer details
Submitted by karelsmits on Mon, 12/17/2007 - 2:59pm.
Hi all,
I am setting up Gizmo, I am able to make outgoing calls, but I can't receive incoming phonecalls. Probably my trunk is not configured correctly, I keep on trying and it still does not work. Please if anyone could help me...
Here are my trunk details for INCOMING (the outgoing is working);
Incoming PEER Details:
allow=ulaw&ilbc&gsm
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
fromdomain=proxy01.sipphone.com
host=proxy01.sipphone.com
insecure=very
secret=YOUR_PASSWD
type=user
user=1740000000
username=1747000000
Register:
17470000000:YOUR_PASSWD@proxy01.sipphone.com
Replace 17470000000 with your own gizmo provided #.
replace YOUR_PASSWD with your own gizmo account password.
Thanks a lot.
Problems with Latest Trixbox
Fri, 05/16/2008 - 4:43am
I'm using the lastest trixbox (2.6.1.0). I can make outbound calls without ANY problems, but inbound calls do not work.
Accodring to the logs Trix is seeing the incoming call, but it is not handling it correctly. It simply sends it to the default 's' and the CDR shows a record of the call.
DEBUG[13762] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
DEBUG[13762] acl.c: ##### Testing 192.168.128.82 with 192.168.0.0
DEBUG[13762] chan_sip.c: Target address 192.168.128.82 is not local, substituting externip
DEBUG[13762] chan_sip.c: Stopping retransmission on '56199a79500937bf0cb0173e4d66eef6@85.145.196.14' of Request 102: Match Found
DEBUG[13762] chan_sip.c: Target address 192.168.128.81 is not local, substituting externip
DEBUG[13762] chan_sip.c: Stopping retransmission on '2e473c921ce24c117fd79e846150f308@85.145.196.14' of Request 102: Match Found
DEBUG[13856] manager.c: Manager received command 'Challenge'
DEBUG[13856] manager.c: Manager received command 'Login'
DEBUG[13856] config.c: Parsing /etc/asterisk/manager.conf
DEBUG[13856] config.c: Parsing /etc/asterisk/manager_additional.conf
DEBUG[13856] config.c: Parsing /etc/asterisk/manager_custom.conf
DEBUG[13762] chan_sip.c: Auto destroying SIP dialog 'TC91oBm8J1IRjAiy@192.168.128.81'
DEBUG[13762] chan_sip.c: Setting NAT on RTP to Off
DEBUG[13762] chan_sip.c: Allocating new SIP dialog for 219288a1-f82c-4a68-b825-9dad0f7a05cf:cwAttempt0 - INVITE (With RTP)
DEBUG[13762] chan_sip.c: Setting NAT on RTP to On
DEBUG[13762] chan_sip.c: Checking SIP call limits for device 174760*******
DEBUG[13857] pbx.c: Launching 'Playback'
DEBUG[13857] chan_sip.c: SIP answering channel: SIP/17476******-08c8e1e8
DEBUG[13857] chan_sip.c: Setting framing from config on incoming call
DEBUG[13857] channel.c: Set channel SIP/17476******-08c8e1e8 to write format slin
DEBUG[13857] rtp.c: Ooh, format changed from unknown to ulaw
DEBUG[13857] rtp.c: Created smoother: format: 4 ms: 20 len: 160
DEBUG[13857] channel.c: Scheduling timer at 160 sample intervals
DEBUG[13762] chan_sip.c: Stopping retransmission on '219288a1-f82c-4a68-b825-9dad0f7a05cf:cwAttempt0' of Response 1: Match Found
DEBUG[13857] rtp.c: RTP NAT: Got audio from other end. Now sending to address 206.81.178.67:6484
DEBUG[13857] channel.c: Scheduling timer at 8 sample intervals
DEBUG[13857] channel.c: Scheduling timer at 0 sample intervals
DEBUG[13857] channel.c: Scheduling timer at 0 sample intervals
DEBUG[13857] channel.c: Set channel SIP/17476******-08c8e1e8 to write format ulaw
DEBUG[13857] pbx.c: Launching 'Macro'
DEBUG[13857] pbx.c: Launching 'ResetCDR'
DEBUG[13857] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
DEBUG[13857] 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,uniqueid,userfield) VALUES ('2008-05-16 13:20:41','+1512*******','+1512*******','s','default', 'SIP/17476*****-08c8e1e8','','ResetCDR','w',1,1,'ANSWERED',3,'','1210936841.3','')
I did some tweaking and I was at one point to get the calling end to hear rining, but on the asterisk end, nothing rang. I've played with the inbound routes in the WebUI, but no luck.
Any help would be appreciated.
thanks,
James
DOOOHH!!!
Fri, 05/16/2008 - 6:05am
It turned out it was the context for the "OUTGOING" Peer which I needed to change. Once I changed this. I went to /etc/asterisk/extensions_custom.conf. Created a new stanza for this context (i.e. from-gizmo):
[from-gizmo]
exten => s,1,Goto(ext-group,6001,1)
or
[from-gizmo] ; Tried this first
exten => s,1,Dial(SIP/xxxx)
Both worked
I then restarted asterisk and my phone's rang.
(I've been trying to get the incoming context right for a while. Since the Gizmo website says what the outgoing context is, I was completely ignoring it).
I found some useful instructions @
http://users.757.org/~joat/wiki/index.php/Asterisk_and_Gizmo
When I tried looking at why mine didn't work, I noticed that it was the outgoing context I needed to change and BOOM, it works.
Member Since:
2007-10-12