[asterisk-users] dtmf Caller-id detection before first ring

2011-05-28 Thread Ashik Ali
Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear cal

Re: [asterisk-users] Audio dropping

2011-05-28 Thread Mark Scholten
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 27 May, 2011 10:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audio dropping On Fri, 2011-05-27

[asterisk-users] Cisco registration problem with 1.8.3.3

2011-05-28 Thread Ian S. Worthington
I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be

Re: [asterisk-users] Cisco registration problem with 1.8.3.3

2011-05-28 Thread Ryan Wagoner
On Sat, May 28, 2011 at 4:08 PM, Ian S. Worthington wrote: > I am having a problem registering my cisco phones which is exactly like that > described in > > http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html > > except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-

Re: [asterisk-users] Cisco registration problem with 1.8.3.3

2011-05-28 Thread Ian S. Worthington
I too had heard that 1833 did NOT have the 184 problem, which makes me suspicious that it's not that. I don't think its a NAT problem. Neither a sip trace not tcpdump show any response at all to the incoming REGISTER. The phone is on the local lan. I have nat=no and nat_enable: "0" i -

Re: [asterisk-users] Audio dropping

2011-05-28 Thread Roger Burton West
On Fri, May 27, 2011 at 10:31:57AM +0200, Mark Scholten wrote: >What could the reason be audio in 1 direction is dropping? (Normally from >the Asterisk server to the mentioned SIP clients.) No clear information is >in the logs (it is like the call ended normally) and not all calls are >having prob

Re: [asterisk-users] Cisco registration problem with 1.8.3.3

2011-05-28 Thread Ryan Wagoner
On Sat, May 28, 2011 at 5:18 PM, Ian S. Worthington wrote: > I too had heard that 1833 did NOT have the 184 problem, which makes me > suspicious that it's not that. > > I don't think its a NAT problem.  Neither a sip trace not tcpdump show any > response at all to the incoming REGISTER. > > The ph

Re: [asterisk-users] dtmf Caller-id detection before first ring

2011-05-28 Thread Pezhman Lali
you have to do these: 1-find suitable patch for your driver(wctdm.c) where the cidbeforering will be defined. 2-modify the chan_dahdi.c in asterisk, change res to 4000 or higher 3-recompile your driver and asterisk 4-set cidbeforering=1 and cidstart, in the config of dahdi. 5-restart your ma