search for: avio

...sorry, I misunderstood you (and arping -D) This was the result of arping on another host; I thought I should see 2 responses in case of an ip conflict. Arping on the troublesome server gives 0 responses. I just tried with a physical console on that server and there I got the same unresponsive behaviour. Does this rule out network related problems? Mark (m.roth) suggested the vms eating up the video bus. (2 vms with an Oracle database) But I'm not sure how I could test that. Patrick

...1 1.92 | avg5 1.97 | avg15 1.61 | csw 229508 | intr 191786 | MEM | tot 47.1G | free 15.9G | cache 519.3M | buff 109.3M | slab 353.3M | SWP | tot 7.8G | free 7.3G | | vmcom 31.8G | vmlim 31.3G | LVM | g_15k-lv_15k | busy 0% | read 1 | write 98 | avio 0.15 ms | LVM | to-lv_oracle | busy 0% | read 0 | write 66 | avio 0.06 ms | LVM | v_oracletest | busy 0% | read 0 | write 79 | avio 0.05 ms | LVM | uito-lv_root | busy 0% | read 0 | write 1 | avio 3.00 ms | DSK | sdb | busy 0% | read...

Hi. I'm having trouble controlling the user info when sending e-mails from asterisk via sendmail to a Microsoft exchange server. When I receive the email the sender is always "asterisk@TheDomainISpecify.com" and the name of the sender is always "Added by portage for asterisk". I want to change both sender-address and the name of the sender. I'm using Gentoo for my

As mentioned earlier I did try that. Someone suggested that there might be an issue with sendmail not "trusting" the asterisk user. And the default behaviour of that is to not allow modification of the "fromstring" and "serveremail". So if you have any idea how to fix that in Gentoo I would really appreciate it. Thanks -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists...

...gt;> >>> >>> atop seems to indicate that sdd is busy? >> >> DSK | sdd | busy 96% | | read 1455 | write >> 1319 | KiB/r 5 | KiB/w 9 | | MBr/s 0.74 | MBw/s >> 1.26 | avq 1.01 | | avio 3.43 ms | >> >> # mount | grep sdd >> /dev/sdd2 on /vm type ext4 (rw,relatime,seclabel,data=ordered) >> >> > And it doesn't do that in any other process on the host? It looks like > it's not related to virtualisation... > > Hi, I changed from vmdk...

...all. The way around this is to have your sip devices register under different accounts and create a ring group (dial(SIP/dev1&SIP/dev2&SIP/devN)) AFAIK, there isn't a reliable method of determining if a sip device is busy other than calling it. On 5/1/06, Arne Morten Johansen <amj@avio.no> wrote: Hi. How does this work? What if this SIP/account was a member (agent) of a queue? Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy when one of the phones is actively talking, or will the other phones continue to ring? You may have seen my other submissions to thi...

Hey. How would I set up my dialplan if a user wants to call its voicemail from an external phone? I'm thinking of getting the user to enter its mailbox number. Something like this: 1. User calls the dedicated voicemail number. 2. Phone prompts for mailbox number. 3. Voicemail(${mailboxnr}@context) Thanks.

Thanks. I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password: NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received Is this a DTMF failure of some sort? Thanks again. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com

Hi there. We're going to develop a call centre app for internal use in our office. The call centre is probably going to be a java-based client installed on a windows machine that our secretary can use. Features should be a way to see incoming calls, answer them, and then transfer the calls to our different users/groups/divisions. If it also could be possible to have a way to see if the user

...busy or the CPU is busy or something else is blocking it > > atop seems to indicate that sdd is busy? DSK | sdd | busy 96% | | read 1455 | write 1319 | KiB/r 5 | KiB/w 9 | | MBr/s 0.74 | MBw/s 1.26 | avq 1.01 | | avio 3.43 ms | # mount | grep sdd /dev/sdd2 on /vm type ext4 (rw,relatime,seclabel,data=ordered) guest configuration is below: <domain type='kvm'> <name>win10-box</name> <uuid>d4a67adb-5d30-42f8-b8c6-d3c4598700f2</uuid> <memory unit='KiB'>419...

[outgoing-fax] exten => _0XXXXXXXXX,1,SetVar(NumberCalled=${EXTEN}) exten => _0XXXXXXXXX,2,Wait(10) exten => fax,1,SetCallerid(${FAX_CALLERID}) exten => fax,2,Dial(Zap/g1/${NumberCalled},60) exten => fax,3,Hangup exten => t,1,Busy exten => i,1,Busy -----Oorspronkelijk bericht----- Van: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]

Hello. I just emerged spanDSP with all the packages needed. After a bit of configuration i was read to test. But i get this errormessage stating that application rxfax was not found. I could't fint rxfax i teh modules directory. I use asterisk 1.0.7. I did reset the server after emerging SpanDSP I use gentoo kernel 2.6 I don't know what else to do. Regards, Arne Morten

Oh. So how can I do this? If I write something in PHP, how do I make it output to an Asterisk variabel? I need to set a variable in asterisk to TRUE or FALSE based on the result of the PHP-script. ________________________________ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Morgan Gilroy Sendt: 8. februar 2006 15:28 Til:

...>>>> atop seems to indicate that sdd is busy? >>> >>> DSK | sdd | busy 96% | | read 1455 | write >>> 1319 | KiB/r 5 | KiB/w 9 | | MBr/s 0.74 | MBw/s >>> 1.26 | avq 1.01 | | avio 3.43 ms | >>> >>> # mount | grep sdd >>> /dev/sdd2 on /vm type ext4 (rw,relatime,seclabel,data=ordered) >>> >>> >> And it doesn't do that in any other process on the host? It looks like >> it's not related to virtualisation... >&g...

Hi. How does this work? What if this SIP/account was a member (agent) of a queue? Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy when one of the phones is actively talking, or will the other phones continue to ring? You may have seen my other submissions to this list. I'm looking for a way to make the other phones in a group unavailable when one of them is busy. Because

I also want to try that asterisk guide. But i'm not sure if i understood it correctly. What exactly do i need to do? Do i need to compile Asterisk with the spanDSP plugin or just configure extensions.conf? The URL to spanDSP in the guide wasn't working. I also use a traditional internet line to recieve calls and hopefully i will get Fax working soon. This is so confusing. Thanks, Arne

What about faxing yourself if you don't have a scanner? -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Johan van Tongeren Sendt: 5. september 2005 09:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] sending fax [macro-fax-dialing] exten =>

...something else is blocking it >> >> >atop seems to indicate that sdd is busy? > >DSK | sdd | busy 96% | | read 1455 | write >1319 | KiB/r 5 | KiB/w 9 | | MBr/s 0.74 | MBw/s > 1.26 | avq 1.01 | | avio 3.43 ms | > ># mount | grep sdd >/dev/sdd2 on /vm type ext4 (rw,relatime,seclabel,data=ordered) > And it doesn't do that in any other process on the host? It looks like it's not related to virtualisation... > >guest configuration is below: > ><domain type='...

Hi. I've started working on a PHP-project that generates the configuration files i need based on what's in my MYSQL database. I can add, delete and edit users from the web. I can set up exactly the dialplan i need by arranging the users in a firms and groups if needed. I've also set up a java servlet so that i can get asterisk to reload by pushing a button from the web-interface. The

Hi there. Is it possible to have different sip users have the same CallerId number in sip.conf. I need this because we got multiple companies on this Asterisk box. Company A's internal numbers: CID: User: 1000 - User 1 2000 - User 2 3000 - User 3 4000 - User 4 Company B's internal numbers: CID: User: 1000 - User 5 2000 - User 6 3000 - User 7 4000 - User 8 Is this allowed? Regards