similar to: Polycom Shared Call Appearance

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

The Polycom 600 supports Shared Call Appearance Signaling. The Polycom documentation states: "Incoming calls can be presented to multiple phones simultaneously. This feature is dependent on support from a SIP server which binds the appearances together logically and looks after the necessary state notifications and performs an access control function." "The phone supports

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 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

We are currently looking for knowledgeable Asterisk system technicians in the NJ area. Candidates MUST be competent, qualified, 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, installations, or as a member of a project

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 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

I take it SCA/BLA isn't going to make it into 1.4. Anyone have any idea when support will be added to asterisk for this? Sean

Has anyone successfully implemented concurrent appearance of the same PBX extension on multiple SIP phones? When using Cisco 7960s under call manager, you can have several phones with the same line appearance, but the first user to seize a line makes it inaccessible to other phones. Under SIP operation it seems as though this is not possible, but we don't see group ringing definable for

Hallo, I am playing arround with dovecot I hat version 1.0.beta7 running fine. After installing beta9 there is a problem with default_mail_env setting. I have all users in ldap. there is an attribute "mailMessageStore" which contains a relative path to users maildir mailMessageStore: sca/ my mailer completes this relative pathinformation to /var/mail/sca/ therefor I setup

I'm using rsync 2.63 on a NetWare 6.5 server backing up various volumes to a SLES 9 server. My script that I'm using on the NW server is: # Rsync synchronisation of APPS rsync -rRutzvP --volume=apps: ./ 192.168.1.252::SCA/apps # Rsync synchronisation of DATA rsync -rRutzvP --volume=Data: ./ 192.168.1.252::SCA/Data # Rsync synchronisation ofGWMAIL rsync -rRutzvP --volume=GWMAIL: ./

Dear Sir or Ms: My OS is ubuntu 12.04(AMD64). I have installed TexLive 2012, Graphviz 2.26.3, doxygen 1.7.6.1 without the dependence doxygen-latex.The version of LLVM I am working on is 3.2. My configure script is the following: ../llvm-3.2/configure --prefix=/home/me/mywork/sca/llvm/llvm-install2 --enable-targets=host-only --enable-debug-runtime --enable-assertions --enable-doxygen

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040331/c2abf19f/attachment.htm -------------- next part -------------- Hello, Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone tell how to get started implementing it for Asterisk. Regards, Tony

Still looking for some direction with this subject: I think the term is called multi-line appearance.... Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it.... This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to

Hello, I just installed dovecot2-beta2 and configured the lmtp-server + postfix (no virtual users) postfix: mail_version = 2.7-20100117 mailbox_transport = lmtp:unix:lmtp-server lmtp_assume_final = yes dovecot: service lmtp { executable = lmtp protocol = lmtp unix_listener /var/spool/postfix/lmtp-server { group = postfix mode = 0660 user = postfix } } Mail gets delivered to