[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

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,

[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 ianworthing...@usa.net 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

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 problem

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 ianworthing...@usa.net 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

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