similar to: Asterisk connecting to a proprietry PBX

Hi All, In my set-up, I''ve got a cron job that triggers a Puppet run every 20 minutes. I''ve found that on approximately 13 nodes (out of 166), puppetd just hangs. I have to go in, kill the process, remove /var/lib/puppet/state/puppetdlock, and run puppet again and then it''s fine. After a while, it just hangs again so I have to go in, kill the process, etc. Any ideas?

Has anyone been able to successfully integrate the Avaya AUDIX voicemail system with Asterisk? I have been diligently investigating converting our small (Ontario, Canada) office to Asterisk, and ditching our Avaya PBX. However, our head office (New Jersey, USA) maintains our AUDIX system, and a) have no intentions of leaving it and b) some users rely upon AUDIX's ability to transfer

Hi All I've compiled Samba 3.0 alpha 17 using --with-ldapsam. In smb.conf I've got: passdb backend = ldapsam ldap admin dn = "cn=Manager,o=Sambatest,c=AU" ldap suffix = c=AU ldap ssl = off ... and the admin dn password is in the secrets.tdb file. Whenever I try to connect to samba, I get the following error message: [2002/05/28 09:34:07, 5]

Hi All, This topic has come up before in the Asterisk mailing list many times, so I know that a lot of people have given up in waiting for a FXO card to be approved by the Australian telecommunications authority. My question is: all legalities aside - is anyone using a FXO card in Australia successfully? Thanks in advance. Regards, Gonzalo

Hi All, I''ve downloaded a Puppet module and I''m trying to add a parameter to it by editing lib/puppet/type/<resource>.rb. I simply added: newproperty(:pcfree) do desc "My description here" end In the corresponding file in lib/puppet/provider, I do something with :pcfree. Whenever I call the resource from Puppet with my new parameter, it keeps

Hi All, I've recently got a "second" number installed on my PSTN line, trusting the Asterisk distinctive ring detection would work as expected. It appeared to work fine at the start, as the second number generated a different ring pattern to 0,0,0 (in the console) only to realise that almost every phone call to this "second" number generated a different ring pattern.

Hi All, I'm trying to achieve the following: - If <sip/iax user> logs in from home, they can dial internal extensions only (this is to avoid employees going wild on local/mobile calls from home) - If <sip/iax user> logs in from the office, they can call anyone they want. Since I have my users defined in an LDAP tree, I'd like to stick to one-account-per-user (each account is

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin

We've moved within the last two months to Asterisk 1.4.x All remote facilities are connected via highspeed (9mbit) connections (Over OpenVPN) to a central Asterisk box, acting as a voice router, that funnels all calls into our Avaya Definity G3R via PRI. When corporate employees visit the remote facilities and try to call the G3R's voice mail system(Audix), DTMF is not recognized unless

Hi, It''s easy enough to modify, say, the "exclude" variable in /etc/yum.conf with something like: augeas { yum_exclude_kernel: context => ''/files/etc/yum.conf/main'', changes => ''set exclude kernel*'' } However, I''d like a more flexible system where I can "build" the list of excludes from

Hi Everyone, So I have been further exploring the integration of our asterisk server and our lucent definity g3si system. I took the suggestion of setting up an isdn-pri line added the two way tie trunk and the signalling group, but can't seem to get the PRI signalling working on the asterisk correctly. I've set pri type to network on the lucent, and pri_cpe in zapata on the asterisk, but

We have had quite a bit of success with our T100P and TE410P cards interfacing to Nortel Meridian PBXes and also to a Livingston Portmaster 3 using ESF/B8ZS and various combinations of E&M wink and ISDN PRI (usually in 5ESS mode). In the near future, I may also need to interface to a Definity via a T1. I was planning to use PRI--is that an option for you on your switch? -----Original

Hello. I vaughely remember someone talking about an asterisk implementation at a University in germany some months back. Any other information ? Regards...Martin -- Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Free MS-Office replacement for most platforms http://www.openoffice.org/

Hi All, I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree where the users will each have their account, SIP username/password, extension number, context, etc. My first question is: can this be done with 1.4.x? If so, where can I get the res_config_ldap from?? I googled quite a bit and found a res_config_ldap that looks to be coded for 1.2. Is anyone running

This is my first post, so please feel free to direct me to another list if needed. I am in the early stages of researching Asterisk. I administer a small Avaya Definity G3 switch (~400 users). Can anyone point my to resources/documents/actual implementation notes of using Asterisk's voicemail with an Avaya Definity switch? Many thanks, Brian Hudson --------------

Hi everyone, We have been working with Asterisk for a while now and would really like to expand its capabilities by fully integrating it with our Definity g3si and are wondering about other peoples experiences with similar setups. Thus far we have only been able to achieve a partial 1 way integration, but ultimately would love to route inbound voip calls to the asterisk out of the lucent.

Hi, I would like to set two separate asterisks to talk to each other. Any suggestions? I'm a "baby" asterisk fan, only started to play two weeks ago, first managed to use kphone with asterisk and a X100P card that is up and running as well. Thanks, Eugen Find local movie times and trailers on Yahoo! Movies. http://au.movies.yahoo.com

Hi, is there a way to retrieve a VM message pressing some key during the greeting playback? Our scenario is a PBX with analog trunks and no DID. There's a general mailbox and no way to assign a number to voicemail. I've seen these question before in this list, but seen no answer to it... Thanks, Renato

Does Asterisk work with Lucent or any other PBX phone systems????

Hi All, I have a particular requirement where a Puppet managed group needs to have several members that are either local and not managed by Puppet (e.g. mysql) or they reside in LDAP. Apart from running an exec call to "groupmems", is there another way to achieve this? Thanks in advance. Gonzalo -- You received this message because you are subscribed to the Google Groups