similar to: Looking for Full or Part time asterisk techs

We are currently looking for knowledgeable Asterisk system technicians in the NJ area. Candidates MUST be competent, qualified, efficient, expedient, trustworthy, and reliable. Must have deployed a few systems and be very familiar with all aspects of installing and configuring asterisk. The technician must be able to follow instructions as well as, work independently on service calls,

Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

Anyone know why zaptel would ignore a facility message from an ISDN PRI. I am trying to get Callerid name to work. The carrier says it on and I see it in the pri debug but asterisk never gets it. Any help would be appreciated. Thanks John Bittner Simlab.net Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 572/0x23C) (Terminator) > Message type: ALERTING (1) >

Anyone get the Nortel 1140E phones working with Asterisk ? These look like great phones and I would like to start using them on our deployments. I know these will work with Asterisk but the sample config files are hard to find. My next step, if I cant find anything on this list is to purchase a Nortel Communication Server for testing. If anyone has a used NCS that works with these phone via SIP

Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling forward unconditional then Asterisk accepts the call but without cname. Below is a trace. Any help is appreciated. Thanks John Bittner Simlab.net voippbx01*CLI> <-- SIP read from 216.86.35.24:63549: INVITE sip:9734333001 at 69.60.198.130:5060 SIP/2.0

Has anyone got Polycom Shared Call Appearance working with Asterisk ? If Asterisk doesn't support this, I am willing to put up a bounty of 1000 to get it to work. John Bittner Simlab.net Shared Call Appearance Signaling A shared line is an address of record managed by a server. The server allows multiple endpoints to register locations against the address of record. SoundPointR IP

I started working on testing asterisk@home. I have setup the system with 5 phones and 1 pots line. I am using polycom phones for this system. Polycom's register and can make outbound calls with no issues. When I make an internal call... The calls go straight to vm without ringing any phones. Incomming pots call do the same thing. Went crazy thinking it was a polycom config issue...but its

Anyone know why chanspy was not included in asterisk distribution as of October. ? I tried patching my current 1.0 but seems the patches are for an older version. I posted a bounty of $250 to get this to work with the newest stable. Needs be able to monitor bridged sip calls with or without a monitoring beep. Thanks John Bittner Simlab.net

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly

I appologize if this post is a duplicate, but I did not get my first post myself.. Here is my original post: --------------------------------- Hi all, I've got some troubles with a Lattice plot. The error I get is Error in Ops.factor(groups[subscripts], vals[i]) : Level sets of factors are different which sort of doesn't make sense to me. Can you please help me figure out what

Hi -- I'm connecting 64bit win 7 professional to an SGI tezro running samba 3.5.8. I connect fine, and everything works except, I cannot read or copy large (40 Mb) files. This used to work when I ran win XP and samba 2. I had to upgrade to version 3 for win 7 to connect. Any clues would be appreciated. Ellis Golub -- ============================================================= Ellis

I am currently running Samba 3.0.25rc1 in a Silicon Graphics Tezro under IRIX 6.5.30. I was forced to upgrade to Samba 3 in order to connect with Windows 7. Prior to the upgrade, I could access the SGI machine running Samba 2 from Windows XP. After upgrading to Samba 3, I am able to connect, have full access to all my files, and the only problem is that files on the share greater than ~20 Mb

Seems to be all morning. I have not been able to access for several hours now. W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Marcel van Kaam, Fonetica Sent: Tuesday, June 14, 2005 7:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VOIP-INFO down? Hi

PJISP cannot handle the From field when it does not contain a number. Can this be fixed? [Feb 25 12:35:43] ERROR[7143]: pjproject: <?>: sip_transport.c Error processing 400 bytes packet from UDP 8.38.43.67:5060 : PJSIP syntax error exception when parsing 'From' header on line 4 col 40: CANCEL sip:14408785990 at 162.255.138.102:5060 SIP/2.0 Via: SIP/2.0/UDP

hi, i'm trying to set a dovecot/pop + postfix couple with ldap authentification. and i'v got a lots of troubles. hers's my dovecot.conf protocols = pop3 login = imap login = pop3 first_valid_gid = 1000 last_valid_gid = 0 mail_extra_groups = mail default_mail_env = maildir:/var/mail/%1u/%u/Maildir mbox_locks = dotlock flock auth = default auth_mechanisms = plain auth_userdb = ldap

When an agent logs into a queue using AgentCallBackLogin, he should be ready to take calls until he logs out right? For some reason the first time a customer calls the queue, it rings the agent just fine but after the agent hangs up the phone and the next caller calls the queue, no more calls will be transferred to the agent. He shows as logged in, but the calls wait in the queue forever and

[root@asterisk zaptel]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.5-1.315/misc/ztdummy.ko): Invalid module format FATAL: Error running install command for ztdummy [root@asterisk