similar to: Strange Fromuser behavior?

Hello, I've got the following queue.conf: [testQ] music=jr_80 ;Bore the caller with some 80's music announce=queue-testQ ;Announcement to play to the Agent answering strategy=ringall ;Let all hell break lose timeout=60 ;We should answer within 60s retry=5 ; announce-frequenty=15 ;Tell them where the are every 15 seconds announce-holdtime=yes ; Give them

Hi there, I tried to get a few "Optipoint 400 SIP" working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I?m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I

Hi everybody, I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk. It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers. The Optipoint shows "no Server..." (Registrar?) in Display. Sip debug shows no unusual (to me) Messages. Sip show

Hello, I?m looking for Optipoint 420 Standard SIP Firmware to make my first tests with Asterisk and SIP, but I?m unable to find it. Could someone help me? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070415/c2da9cc0/attachment.htm

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

Hello, I would like the dialing party to know what happened to the call, since asterisk doesn't relay a sip error back to the originating sip channel (would be nice, a if (org_channel = sip && dst_channel = sip, relay error to sip client) I want to set up audio feedback on the call status. I've changed the county setting to NL in indications.conf and created this test

Hello, I'm trying to implement a function available on the PSTN net here, if you dial a number which is busy and you press 5, you will be called back when the busy party hangs up. Figuring out if a SIP user is busy isn't to hard, ${DIALSTATUS} produces a BUSY message, however, how can I implement the call back? IE, I dial to extension 712, but that extension is busy, I dial 5 and

Anybody using Siemens optiPoint 300 H.323 phones? I saw a few references to them in the archives of this list, and the Wiki seems to be down. I have a chance to pick up a bunch of these, cheap. Questions: * Asterisk support? * What sort of power supplies will they need? The bunch I am looking at are surplus and have no supplies. Thanks, </edg> Ed Greenberg San Jose, CA

Not strictly a problem with Asterisk but one of my phones. A couple of days ago I decided to update the firmware in my Optipoint 600 Office which looked as though it went swimmingly until that is, it rebooted. Since then the phone just boots up and displays the following: Can't Boot!! Development shell active. It doesn't try to request a DHCP address, in fact it does seem to do

Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? IE, the following setup: SIP Phones -------------- Asterisk ------------------------ SIP carrier | | ------- Asterisk (standby) ------ Asterisk is used as a PABX in this setup, so the

Hi all, I have an optipoint 400 Sip phone with 2.46 sip image which stores its music on hold file localy (on the phone) rather than indicate to the server the call is on hold. i'd like to have Moh play from the server, and i'm not sure if there is a work around to achieve this anyone else has the same phone and came across similar problems ?? thanks fam

List, I've got 2 queues with 10 agents in both queues. One of the agents is mainly responsible for queue_1, and the others mainly for queue_2 so i've defined the following in my queues.conf [queue_1] strategy=ringall member=>Agent/1,2 member=>Agent/2,1 member=>Agent/3,1 member=>Agent/4,1 [queue_2] strategy=ringall member=>Agent/1,1 member=>Agent/2,2

I want to let Asterisk read its configuration from a mysql database. I configured everything according to the wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config. However it doesn't work. I am using 1.0.8 asterisk version and here are my config files: Extconfig.conf: [settings] ;uncomment to load queues.conf via the db engine. ;queues.conf =>

Hi list, first evening of playing with rails, so please forgive me if I ask something stupid. ;-) I created a User model and tried to use ActiveRecord callbacks to convert the password to sha1 just before saving it. For some reason postgresql gives me a error because the given password is null. To test even further I tried to change :login too, same error happens, :login is empty too. I am sure

Hi everybody, I'm seeking information on how to report an IP phone on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a notification to a phone B on the call that is going to A, but this notification is displayed on the B phone display and the user does not need to hit anything to view the information. I'm

Hi, I'm using asterisk 1.2.1 on a debian sarge distro. I've followed notes in http://www.voip-info.org/wiki/view/Polycom+auto-answer+config and http://www.voip-info.org/wiki/index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO but I still cannot change ring style via asterisk using exten => 666,1,SipAddHeader(ALERT_INFO="ring3") in extensions.conf . Is it

Hello, I'm using asterisk to relay sip to another SIP provider, I've setup a friend in sip.conf for my softphone and a user & peer section for the SIP provider, when my softphone calls out to the sip provider and the sip provider returns an error (404 Not Found for example) the sip message is not relayed back to my sip phone, it just sits and waits for a timeout. Is it possible to

I'm just dumping all I got in here, after initial feedback we can see how to organize todo's around it. tl;dr: * Does not upgrade cleanly from 4.8 packages, so we have to prevent this from entering testing until we fix that. * Live migration is broken, explodes with memory allocation errors. ---- >8 ---- 1. Build packages * I have built salsa/master using pbuilder targeting sid.

Hi all, I have some OptiPoint400 standard SIP connected on Asterisk. They work pretty good. I only notice that calling from OptiPoint to OptiPoint doesn't show me the Caller ID name (only Caller ID number). But calling from an OptiPoint to a SoftClient (e.g X-Lite) shows me both on the softclient. Any ideas? Thx! sip.conf [2005] type=friend callerid="OptiPoint" <2005>

Hello Im just been working with puppet, the first usecase i have is to set up a system to create users and then push ssh keys on this machines in the create users .ssh files. I just started with puppet so i am a bit cluesless, Can someone push me in the right direction? Im having a centos enviorment. -- You received this message because you are subscribed to the Google Groups "Puppet