similar to: dialing with different speed

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

Hi, I've got two pa1688 phones that I want to set up to communicate between branch offices without a gatekeeper. Both phones will be behind a firewall and I want to use port forwarding so the phones can communicate. I tested the phones behind a firewall on the same network segment and there were no problems at all using sip. However, I then moved the phones into situ and port forwarded

Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. I do not have a good knowledge of Asterisk, I'm not very familiar with its configuration and I've a

Hello Everyone, I am trying to change our /etc/sudoers (using visudo) to allow 2 commands to be run as root without a password, but it isn't working. Here is the part of the sudoers file that is in question. # User alias specification User_Alias FULLACCESS = doug, scott # members of the FULLACCESS User_Alias may run chown and chmod without a password FULLACCESS ALL = (root) NOPASSWD:

Greetings, I'm having a job getting asterisk to register with a Huawei softX3000 softswitch via SIP. I keep getting 401 Unauthorized. Funny thing is I can successfully register SJPhone, a PA1688 IP Phone as well as a WiFi Phone against the switch without *any* problems. I think it's got to be something as simple as perhaps the register string which is currently

Hi everyone, I am trying to connect 2 asterisk servers via IAX, but it just fails to do so.. I'm using SIP to connect the IP phones on the LAN at the 2 different physical locations where each server resides and I'm able to communicate on my LAN via SIP without any issues. The problem is that I'm unable to make Asterisk servers talk with each other via IAX.. Here is my issue.

I'm just wondering if anyone knows the story with these... http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5721202362&ssPageName=STRK:MEWA:IT He claims they support IAX2 and SIP... but almost no history on the account selling them. I didn't see anything in the wiki about this company either.. Does anyone have any history with these phones? Thanks, Jared

hello i finally have success with winbind!!!! :) thanks everybody so, i use this acl's in squid external_acl_typqe NT_global/-group %LOGIN /usr/local/squid/libexec/wb_group acl FullAccess external NT_global_group "domain group" httpd allow FullAccess httpd deny all with this configuration all is DENID by squid. what???????? it's possible to write acl's specifics for

Hi, I have two Asterisks identically installed on two computers. One of them is directly connected to the Internet, the other one through a NAT router (Netgear MR314). On the one behind the router I have an X100P card installed for PSTN connections. In the local LAN of each PBX they are several hardware IP phones (Cisco 7960 and 7940 with SIP images, firmware image P0S3-04-4-00.bin). I have

I received a sample IP/Speakerphone from my friends in China today. Asterisk setup was fairly uncomplicated and I had it running as an extension on my server within a few minutes. Sounds quality of both the receiver and the speakerphone are fine (wife's opinion). Are there any tests I should run with this phone? Following are the specs: - Single line appearance - Alpha display, 2x16 chars -

I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid="Conference Room Polycom" extensions.conf exten => 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten => 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper

Hi, I'm running samba-3.0.14a_1,1 under FreeBSD 5.4-STABLE. I migrated all my data from samba 2 to samba 3, and I'm having a problem with some users. For example, user john can login perfectly to the network. It?s ok. But user smith not, him get the error ?The network name cannot be found?. But all configurations of john and smith are the same. I found this on Samba Official Hanbook:

Hi, Moving to a new computer (Windows 7) but for reasons of reproduceability I would seriously like to also install my present R 2.8.1 along with all extension packages on that machine as well (that is besides R 2.14.1). What's the best way of doing so? My idea is: Get the setup for 2.8.1 and install it; during install neither select 'save version number in registry' nor

I am about to buy AT-320EE phones for use together with asterisk. Has anyone used these phones ? Are there any known problems using them using Asterisk ?

Hello, I am experiencing weird problem in Asterisk 16.2, possibly a bug. Same thing works fine in Asterisk 11. Here is the situation: I have 2 extensions on 2 phones. 4 extensions in total. phone 1: 8882 8382 phone 2: 8884 8384 And I have 2 SIP trunks for outgoing calls. I want to call via SIP1 when called via 8882 or 8884, and SIP2 when called via 8382 or 8384. And one last detail. SIP1

Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed file, but still get that error and the macro cannot continue. Why? I have tried even with a simple

Hi, Whilst trying to get this Tecom phone working with Asterisk, it seems to be unable to login. Using the 'sip debug' command from the CLI does not produce any output even though the debug of the phone shows it trying to login every second or so? The phone seems to be based on a "Centrality PA1688" processor. Is there any other way I can see why this phone fails to login? I

Hi all, I'm testing the re-invite with an * box and two phones (ywh500 ewith pa1688 chipset) but it doesn't wotk. The phones support the re-invite, and i use this row to dial (without Ttr) exten => s,22,Dial(${tjext}) and put canreinvite = yes in my sip.conf. Anyone can help me pls ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Who else makes Analog Telephone Adapters with IAX protocol besides Digium? I've seen Farfon is advertising their unit but I'm not sure if they shipped it or not. Why is so hard for others to support this protocol in their adapters? SIP is totally unsuitable for firewall traversal for security reason. I would like to get min. two port unit. Is it possible to use IAX protocol without

Hi, i need an recommandation for an ATA adapter to use with an anlog fax maschine. I would appreciate any hints. Regards!