Hey asteriskers!
I know that may look weird, but it's happening:
We have an * server running in a wireless(cellular) operator for IVR services, we bill them per minute, but there is a remarkable difference between our CDR records and their billing system.
* server have a Sangoma, and 3 PRI_ISDN E1 lines are connected to it, * is working perfectly with no problems at all. We requested a re-analysis and they confirmed their records are correct and error free!!
At last we agreed on setting a DB that collects call information at their postgres DB that I post in, BUT I'm having a problem detecting hangups, as I tried DeadAgi on h extension, but the problem is the channels are destroyed and I can't do anything.
Anyone have any suggestions?
Truely/
Joe
From: Jean-Yves Avenard <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion<[email protected]>
To: [email protected]
Subject: [Asterisk-Users] Uniden UIP200 and Asterisk v1.2.4: problem notregistering
Date: Tue, 7 Feb 2006 02:45:15 +1100
Hello
We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since.
We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline:
It does show up in asterisk a few seconds after the UIP200 reboot:
-- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200
but after about 5s I will get something like:
UIP200 is now unreachable.
htpc*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
uip200/uip200 192.168.10.104 D 5061 UNREACHABLE
I have tried the latest firmware (v4.70) and the previous one we've been running for over 18 months (v4.59) without any luck
Here is the sip.conf I've created on a test server where Asterisk is using the port 5061 , same for the UIP200 using port 5061. There is no NAT, the UIP200 is on the same subnet as the asterisk server:
(I'm trying to isolate the issue without affecting our main asterisk server)
[uip200]
type=friend
port=5061
secret=uip200 ; password for registration
nat=never ; phone may be behing nat
host=dynamic
reinvite=no
canreinvite=no
qualify=3000 ; send udp every 2 seconds, to keep nat open
callerid="Jean-Yves <200>"
dtmfmode=rfc2833 ; DTMF mode
context = jya-in ; Default context for incoming calls
disallow=all
allow=ulaw
allow=alaw
allow=g729
If I unable: sip debug ip 192.168.10.104 (the UIP200 IP address), I get every 3-4 seconds on the console:
---
Retransmitting #2 (no NAT) to 192.168.10.104:5061:
OPTIONS sip:[EMAIL PROTECTED]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5061 ;branch=z9hG4bK08dcd4a8;rport
From: "asterisk" <sip:[EMAIL PROTECTED]:5061>;tag=as67a81892
To: <sip:[EMAIL PROTECTED]:5061>
Contact: <sip:[EMAIL PROTECTED]:5061>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 06 Feb 2006 15:40:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
Any help would be greatly appreciated.
Thank you in advance.
Regards
JY
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