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...e person managing the list at > ??? ocfs2-users-owner at oss.oracle.com > > When replying, please edit your Subject line so it is more > specific > than "Re: Contents of Ocfs2-users digest..." > > > Today's Topics: > > ???1. Disk access hang (Gabriele Alberti) > ???2. Re: Disk access hang (Sunil Mushran) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 10 Mar 2010 11:22:34 +0100 > From: Gabriele Alberti <gabriele.alberti at pg.infn.it> > Subject: [Ocfs2-users] Di...

Is there a way to have extensions automatically created for registered sip users ? I did some investigation and found some hope in chan_sip with relation to the somewhat undocumented usereqphone option but i may be totally off track. All i want to be able to do is send a call to number@ip_address where the number is the username configured on the phone that has registered with asterisk

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

Is there a reason the variable ${DNIS} does not get set with incoming calls via chan_capi ? Is it related to the MSN=X in capi.conf ? version = chan_capi-cm-0.6.3 example; exten => _95555555XX,1,NoOp, ${EXTEN}, ${DNIS} == ISDN1: Incoming call '0400000000' -> '95555555' -- Executing SetCDRUserField("CAPI/ISDN1/955555 55-135", "Incoming") in new

Hi all, I am configuring asterisk in a cable modem network, using a motorola TM401A. I can make calls from the MTA but I can receive, display the following error: -- Executing [1500 at alberti:1] Dial("OSS/dsp", "MGCP/aaln/1 at 0-13-11-82-bd-a.ssw.intercal.net|30") in new stack [Sep 2 00:10:53] NOTICE[28062]: chan_mgcp.c:3572 mgcp_request: Asked to get a channel of unsupported format '0' [Sep 2 00:10:53] WARNING[28062]: app_dial.c:1191 dial_exec_full: Unab...

I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco

...- Ciao Nico ----- Messaggio originale ----- Da: "Tom Eastep"<teastep@shorewall.net> Inviato: 07/06/04 20.27.28 A: "Mailing List for Shorewall Users"<shorewall-users@lists.shorewall.net> Oggetto: Re: [Shorewall-users] Re: Proxy arp users Nico Alberti wrote: > > But from an external interface I can ping both 80.23.252.56 (network) > than 80.23.252.57 (telco router). > When you ping the network, do you have to use the -b option to ping? -Tom -- Tom Eastep \ Nothing is foolproof to...

We currently have an office which has a happily working Asterisk setup and all is good. We are looking to deploy the same solution to our other 5 offices, my question relates to the interconnections between all those sites. The requirement is seamless dialing of extensions and the ability to have members of multiple offices participate in call queues. I wish to keep the configuration as

I seem to be having a problem with B3 on my ISDN line, as you can see from the dial string I am having to have asterisk generate ringing else there is no progress indication. -- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/g1/142392203000/ b|40|r") in new stack -- Called g1/142392203000/b -- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/

As I understand there is provisions for hold in both RFC 2543 (SIP) and RFC 3264 (SDP) and asterisk users the latter. The RFC 2543 method tells the UA its media stream is at 0.0.0.0 where as the method via SDP can tell it to listen to any address/port/ protocol combination, which is how asterisk tell it to listen to the audio stream it presents when it is asked to hold a call. Please

I hope this isn't considered cross posting, i sent the following email to Digium support but figured someone on the list may also have better insight into my questions. I have purchased 2 g729 licenses from Digium for testing and have the following questions; ** My configuration is a single asterisk box configured with 2 g729 licenses and 2 x Cisco 7960 Phones, I have confirmed the

Is it normal for the following to occur hourly on an E1 PRI ? -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 --

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I'm having a few issues with the MySQL realtime configuration in CVS-HEAD. I tested it initially with realtime extensions (realtime_ext => mysql,asterisk,extensions) and a realtime switch in extensions.conf and that works fine, So I though I'd go back and test a static configuration mapping. I used the table structure from the asterisk guru postgres howto to create something