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SIP Trunk monitor

kapamarou
Posts: 24
 Member Since:
2010-11-01
Submitted by kapamarou on Fri, 07/29/2011 - 6:57am.

Hello,

i am trying to register a SIP trunk to my provider. The last few days i had many problems (DNS, authentication) which i think are solved. I assigned the trunk to an inbound and to an outbound route but nothing. Is there any way to monitor the trunk as well as the entire call flow?

I had a look in the PBX Status and i can see two SIP peers which are my SIP phones (status OK). One SIP peer which my trunk (status unmonitored). I see no SIP registration.

Thanks

Demetris

‹ Unlimitel - Multiple Trunk Registrations failing Pbx to pbx route inbound ›


stanjohn
Posts: 73
 Member Since:
2011-01-06
Add "qualify=yes" as the
Fri, 07/29/2011 - 4:25pm

Add "qualify=yes" as the last line in the trunk setup, Outgoing Settings, peer details.



kapamarou
Posts: 24
 Member Since:
2010-11-01
Thanks for the
Sun, 07/31/2011 - 11:03pm

Thanks for the reply.

qualify entry made things better. I can see the peer as monitored however i am not able to see the SIP registry. Why?
In addition i am not able to make inbound or outbound calls from the specific trunk even if i attached it on an inbound and outbound route.



kapamarou
Posts: 24
 Member Since:
2010-11-01
TRUNK Dial failed due to CONGESTION
Sun, 07/31/2011 - 11:32pm

I had a look on my asterisk/full and noticed the error

TRUNK Dial failed due to CONGESTION

what does this mean?

i had a look on
Asterisk CLI
sip show peers Pilot

* Name : Pilot
Secret :
MD5Secret :
Context : from-sip-external
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : ""
MaxCallBR : 384 kbps
Expire : -1
Insecure : port,invite
Nat : RFC3581
ACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : xxxxxxxxx
Addr->IP : xxx.xxx.xxx.xxx Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Transport : UDP
Def. Username: username@xxxxxxx
SIP Options : (none)
Codecs : 0x28000c (ulaw|alaw|h263|h264)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing : No
100 on REG : No
Status : OK (32 ms)
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs

What do i do wrong?



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