similar to: Trasnfer to a different VM box after leaving a VM

Any suggestions to look into, or more debug info required? The box is running centos 4.3 final, up to date with the latest versions of CentOS updates,, it's running something like 3.0.10 or .11 - can't recall. 1:22 am. Off to bed... Thanks. -----Original Message----- From: samba-bounces+dmacpherson=mainframe.ca@lists.samba.org

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify?

We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on "attended transfers". What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to

Is there a way do what is sometimes called a 3rd party transfer in Asterisk. That is; Call A comes in and is answered B. B then places A on hold and calls C. After C answers, B&C chat for a moment, then B brings A on line. After making intro's B then drops off call. Thanks, Jeff NeturallySpeaking -------------- next part -------------- An HTML attachment was scrubbed...

More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30

Hi, I'm having prolems with attended transfer configuration. Does this feature had been implemmented by any of you??? What is the best * version to make this work?? Some sample example?? I'm using like this in features.conf: [featuremap] atxfer => 900 ; Attended transfer My * version was intalled by *@home CD and is 1.0.9. Thanks in advance.

i'm having an issue when transferring large files to the samba servers from an xp client - files about 1GB or larger. about 70% into the transfer i get a network share no longer exists error and the transfer fails. I can pull down from the server fine with no issues. The XP machines are authenticating from a different Samba server though the problem is with that machine as well. Error and

Has anyone seen this (version 1.2)? The following dialplan should result in the voicemail message being delivered two both mailboxes (5555557426@context_name and 5555557427@context_name); exten => 0,1,SetCIDName(OPER ${CALLERIDNAME}) exten => 0,2,Dial(SIP/5555557423&SIP/5555557424&SIP/5555557425&SIP/5555557426&SIP

Hello All, I am using ODBC storage for voicemail on my asterisk box. I want to have a common voicemail box for different extensions. I know how to do that, but the question troubling me is how and where do I store the the extension name for which a particular voicemail was left. e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 55555. Now, when someone calls 1000, and leaves a

Hi, it seems if a user leaves voicemail and hangs up the call when done, then HasNewVoicemail never gets called on the next line in the context. However if they press # to finish their VM, then it moves to HasNewVoicemail and this works? eg:- ...... exten => 2002,3,VoiceMail(u${OFFICEVM}) exten => 2002,4,HasNewVoiceMail(2002) ...... exten => 2002,105, do something cos vm has been

Hello, I have started having a strange problem. Asterisk is connected via 4 analog lines to PSTN and we have SIP phones internally. All was working fine but recently each time a user calls from PSTN and when he is leaving a voicemail for someone, the caller gets disconnected after 30 secs. We have AMP installed. This is reproducible and is happening always. It seems that Asterisk is disconnecting

Hi, I''m trying to build an ipsec gateway and somewhere I''m doing something wrong. I have a couple of routers that have clients in their back. All the routers are connected into a switch. In that switch I also have a computer that provides internet access to the clients. I would like to setup some sort of autentification (don''t need encryption), to allow me

Hi. Has anyone experienced with the new firmware .77 ? There's Day Light Saving time now, but haven't time to play with it, till now. Matteo. -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : mbrancaleoni@espia.it

This is mainly for testing the new parallel python threading model, but it also an example how to manage multiple connection from a plugin. I tested this with local imageio server, serving qcow2 image on local SSD. Start imageio server from imageio source: ./ovirt-imageio -c test Create test disk: qemu-img create -f qcow2 /var/tmp/disk.qcow2 6g Add ticket to accessing the image,

Hi guys, I'm a photographer and I want to use the software that my camera manufacturer makes. I can install the application like View NX 2 but Nikon trasnfer makes an error. I want to know what information do I have to gather for help or what should I need to correct this erros. If any interested of helping please just reply. Thank you.

All, - 1 call transfer - Call comes in for uip200. can trasnfer it just fine. - 2 call transfer - Call comes in - then a second call comes in I can longer transfer either call? I can toggle between them but not transfer. Does anyone know how to acomplish this? jerry

Hello list, I have samba 4 as BDC,I need to promove this server as PDC and eliminate my windows server 2003, but I do not want to write all users again, then I can make it as follow: 1- In the console of windows run ntdsutil 2- After write roles and press enter 3- write connections and press enter 4- write connect to server \\PC with samba4 5- write quit and press enter 6- Write in this order

moving to a temporary server could help? what i mean is: trasnfer fsmo roles to samba02 join temp01 demote old samba01 bring up new samba01 join samba01 remote temp01 correct? this should work? Am 27.10.2016 um 13:09 schrieb Rowland Penny via samba: > On Thu, 27 Oct 2016 11:10:58 +0200 > Dirk Laurenz via samba <samba at lists.samba.org> wrote: > >> Hi, >>

Hi list, in asterisk 1.4 and maybe earlier it was possible to use voicemail system with mailboxes starting with some special characters like *. The line in voicemail.conf was like this: *123 => , AB,,,tz=cet|attach=no| Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4. In Asterisk 1.8 the above scenario is not working any more. The Voicemail application reports an

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello list,</p><p><br></p><p>I am asking when a caller want to leave a message to a mailbox with the application voicemail</p><p>How i can limit the duration for exemple 30 seconds.</p><p>exten =>