similar to: Paetec SIP Trunk

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

Are there plans to expand the logging capabilities in OpenSSH, so that the details of what files were moved using sftp is included? If not, does anyone know of a good way to capture this information? Thanks in advance, - Sue Susan K. Diller UNIX Systems Administration PAETEC Communications, Inc. 600 WillowBrook Office Park Fairport, New York 14450 *(585) 413-2320 * susan.diller at

Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge of DYI wiring across remaining buildings. Phones - Total of about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will have to be analog due to the distance.

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi, Asterisk 1.4.17 Sangoma a102DE I'm having some issues sending CallerID Name to a Dialogic based phone app. According to the pri debug (asterisk2a-pri-debug.txt in [3]) you can see that it is sending the CallerID Name "Mike - Budgetone - reachme.com" to the Dialogic card, but it isn't regestering on the Dialogic based system. I can receive CallerID Names from our Paetec,

I am thinking about a change to our company's phone "layout" and would like to get comments from people who have done something similar. Currently, we have 3 locations - each with their own Asterisk PBX. The corporate office has a PRI. Each remote location has a SIP provider for 5 channels of SIP going to their own PBX. Interoffice calls use the PSTN. Most inbound calls come to

Hello, I've been working on this for over a week. I don't think its working the way it should. Here's what I'm trying to do: I have a Windstream dsl router with wireless > 192.168.1.100 | | Centos 6.8 server eth0 > 192.168.1.110 > gateway 192.168.1.111 eth1 > 192.168.1.111 > gateway 192.168.1.100 | | Switch > other

We have an Asterisk box setup and are ready to start offering VOIP to our Wireless and DSL customers. Who do you guys recommend for DID's, 911, Long Distance, etc.? We are looking for a solution to use where we can provide the Asterisk box and they provide everything else. Thanks, K -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello, i'm having a wierd problem with 0.31 tftpd-hpa. i'm using xinetd, with this config: service tftp { disable = no socket_type = dgram wait = yes user = root log_on_failure += USERID bind = 10.13.0.254 server = /usr/sbin/in.tftpd

Hi I;ve trixbox installed with 2 NICs. One NIC carries the MPLS-VPN traffic (only a 1 MB link without internet for carrying voice to another site) while the other NIC has a connection with public IP for internet services on that machine. the first NIC (eth0) has IP of 172.16.0.1 and is connected to router with WAN IP: 10.18.6.254 , the second IP is 203.234.82.98 (eth1). i want to have the

When trying to migrate my db using rake I get the following error message: rake aborted! No migration with version number 3 I am using sqlite3 as my database server. Any help is much appreciated. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email

Hello, Our Asterisk runs with multiple remote sites (12 over an MPLS network), everything works fine except for the last site we have juste installed. When VOIP flows comes/goes from/to this site, there are sound quality issues, persistent, 100% reproducible, on every call. This is not a bandwidth or latency or jitter problem, everything is fine on the network. Our MPLS provider does all check

Good afteroon all, First of all: thanks for everybody who is willing to think this through with me: I'm having some issues regarding call quality between some calls. Let me try to explain the situation first We have a Asterisk 11.16 server based on the Xivo distribution. There are 2 servers running in cluster (Active Passive), both virtual with the following config: Quadcore CPU 8 GB ram

Just a Question. I would like to know if TDMoE follows specifiaciones of TDMoIP RAD protocol that says that there is a compression of 16/1 when you do TDMoIP. Manuel Marin Garcia TRANSTELCO S.A. DE C.V. Campos Eliseos 9050 B4 – Cd. Juárez, Chih. 32452 - México Oficina: +52 656 692 11 09 – Fax: +52 656 692 1112 - Celular: 915 727 6141 http://www.transtelco.com.mx

Hello, sorry for this little OT post. In my company, we have 2 distant facilities, with people at each facility working on the same files. The 2 facilities are connected through MPLS with about 10MBytes/s BW. Saving the work files to servers located at one or the other facility has became a pain for the people accessing the files from the remote site. Trying to improve files

Hello Shorewall users I have a firewall based on RedHat 8.0 and Shorewall. I have 2 interfaces, with 2 ip address on the loc interface, the connection to the internet runs through my company''s network with an ADSL/MPLS line. I need to configure my Shorewall with the possiblity to deny some users'' access to the ''net'' for some subnet. Ex. my son''s

In my spec_helper, I want to load yaml file to a variable, and then be able to use that in my tests. spec_helper.rb: RSpec.configure do |config| end config = YAML::Load(......) some_spec.rb describe ''blah'' do it "should be...." do MyClass.new( config[''test''] ) end end How can I do this? -------------- next part -------------- An

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

Afternoon All, Just ran yum update and it burped on the following: Package xulrunner.i386 0:1.9.2.20-3.el5_7 set to be updated --> Processing Conflict: firefox conflicts xulrunner >= 1.9.2.19 firefox-3.6.18-1.el5.centos.i386 from installed has depsolving problems --> firefox conflicts with xulrunner Uname -a gives: Linux 2.6.18-274.el5 #1 SMP Fri Jul 22 04:49:12 EDT 2011 i686 i686

Hi All, Quick question, I just finished googling and found the thread that is dealing with the rpmdup_remover.sh script, and thanks to everyone that was involved in this. I followed the following threads: http://www.opensubscriber.com/message/centos at centos.org/11086209.html http://www.centos.org/modules/newbb/viewtopic.php?viewmode=flat&topic_id=17166&forum=37 Which lead me to the