Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread C F
First off, if they are on the same network without any nat, then it is not needed at all. Since this works well with pre 1.2b2 I would say you should open up a but on the bug tracker at: bugs.digium.com. I did not yet update to 1.2bx so I have no way of confirming this. Thank You. On 11/7/05, Wald

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread Waldo Rubinstein
Ok. The keepalives work for other phones, but not the UIP200. I have a bunch of X-Lites, X-Pros, SPA-841s, and UIP200s. It works fine in all but the UIP200 (only in 1.2b2). As far as your questions: 1) They are on the same network and same netmask 2) They are not natted. Let me know what

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread C F
If this is the case. then we now know what the problem is. The keepalives from asterisk to the phones were not working in 1.2b2. The question now is why? Please work with this so that we can troubleshoot this to see if it's a bug with 1.2b2 or not. 1. Is the UIP200 on the same subnet as asterisk? 2

RE: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread Anton Krall
alf Of C F |Sent: Monday, November 07, 2005 9:57 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200 | |The unreachable is the problem. Try adding a qualify=no to |that sip entry. | |On 11/7/05, Waldo Rubinstein <[EMAIL PROT

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread Waldo Rubinstein
I do have qualify=yes pretty much in all my sip entries. I just changed all the entries where I have a UIP200 to qualify=no and now they all work. The funny thing is that it worked with qualify=yes in 1.0.9 and 1.2b1 Thanks, Waldo On Nov 7, 2005, at 1:29 PM, C F wrote: I guess that somew

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread C F
I guess that somewhere in your settings you have a qualify on, or that 1.2 has it on by default. Do the following: cd /etc/asterisk grep ".*qualify.*" ./* and see the output, if the only line that has qualify is that qualify=no, then this looks like a bug to me. Please report back. On 11/7/05, Wal

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread Waldo Rubinstein
Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1. Very strange. Anyway, thanks. - Waldo On Nov 7, 2005, at 10:57 AM, C F wrote: The unreachable is the problem. Try adding a qualify=no to that sip entry. On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: Additionall

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread C F
The unreachable is the problem. Try adding a qualify=no to that sip entry. On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: > Additionally: > > *CLI> sip show peer 100074 > > * Name : 100074 > Secret : > MD5Secret: > Context : qa > Subscr.Cont. : > Langua

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread Waldo Rubinstein
Additionally: *CLI> sip show peer 100074 * Name : 100074 Secret : MD5Secret: Context : qa Subscr.Cont. : Language : en AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PR

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread Waldo Rubinstein
sip.conf: [general] allowguest=no bindaddr=0.0.0.0 bindport=5060 callevents=yes defaultexpirey=300 externip=204.74.89.12 externip=204.74.89.13 localnet=10.0.10.0/255.255.255.0 maxexpirey=3600 relaxdtmf=yes srvlookup=yes tos=lowdelay videosupport=no ; global channel settings disallow=all allow=ul

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-06 Thread C F
can you post the sip.conf for that uip200? On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: > When I dial the extension, I get this: > > -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20") in new > stack >== Everyone is busy/congested at this time (1:0/0/1) > > > When I do a

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-06 Thread Waldo Rubinstein
When I dial the extension, I get this: -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20") in new stack == Everyone is busy/congested at this time (1:0/0/1) When I do a sip show peer 100074, everything it shows matches the results of executing the same sip show peer on * 1.0.9

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-06 Thread C F
Whats the exact CLI output you are getting when calling that extension? On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: > Nope. It isn't active. I even factory reseted the phone but still the > same. One more piece of information: it works just fine in 1.2b1. I > beginning to think it coul

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-05 Thread Waldo Rubinstein
Nope. It isn't active. I even factory reseted the phone but still the same. One more piece of information: it works just fine in 1.2b1. I beginning to think it could be a bug in 1.2b2. Any other ideas/suggestions? Thanks, Waldo On Nov 5, 2005, at 9:10 PM, C F wrote: You sure that the DND (

Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-05 Thread C F
You sure that the DND (Do Not Disturb) button is not active on the UIP200? On 11/4/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: > I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro > phones. > > All phones register fine with * and I can place outbound calls with > no problem. > >