similar to: Failed to bind to 0.0.0.0:5060: Address already in use

I have Fedora Core 1 workstations configured with winbind logging onto our NT domain. Everything is going well except that I cannot seem to figure out how to connect to a windows home directory on a Win2000 server. Here is the scenario: W2K server with home directories for each user: Each user has modify permissions to their share Administrator has full control First off, I need to somehow

Hi, I've downloaded and installed the samba 3.0.2 binary packages for Debian 3.0. The problem is, that if I do 'wbinfo -g' I only see a small number of groups, about 10, where there should be almost 30 groups. The 'wbinfo -u' works fine, I see all my users. Could it be that there is a problem with german 'Umlauts (???)'? I have some groups with these characters in

Hello all, I have a quick question regarding SAMBA and the way it handles login scripts. Currently I am running a small network at home consisting of a FreeBSD server running SAMBA 3.0 and I have multiple Windows XP clients on the network. I have configured login scripts with SAMBA, however they are not currently executing during a login. Now, my question is, is it possible to have SAMBA

Platform: RHEL 3 Version: 3.02a For some reason whenever I try to join our domain by using either RH's binary or a custom compiled src.rpm, I get the following: [root@bonair samba]# net join -U dchait dchait password: realm must be set in in smb.conf for ADS join to succeed. ADS join did not work, falling back to RPC... [2004/03/10 12:29:19, 0] lib/util_sock.c:read_socket_with_timeout(279)

Hi, I' still fighting the problem, that I can talk from one SIP phone to another, but I can't hear the output of the playback or similar applications: exten => 202,1,ANSWER() exten => 202,2,PLAYBACK(tt-monkeys) exten => 202,3,HANGUP() When I dial 202, asterisk show the following on the cli: -- Executing [202 at local:1]

Hi, the german telco Colt Telekom has assigned the phone number block 56830-xxx to one of our customers. In the diaplan we have setup extensions like the following ones: exten => 56830910,1,Answer() exten => 56830910,2,Dial(SIP/bduerring,10,tr) exten => 56830910,3,VoiceMail,u20 exten => 56830910,4,hangup exten => 56830910,103,VoiceMail,b20 exten => 56830910,104,hangup exten

Hi, after I fixed my problem with the playback() application, I now have the next strange one. When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If some in the town of this client calls him, he gets the ISDN error "service not available". Out office is connected to he office

Hi, I have a strange problem with our Asterisk installation. Outgoing calls are handled by the following lines: exten => _0[2-9]X.,1,Set(CALLERID(num)=09999403${CALLERID(num)}) exten => _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} = 0999940321]?099994030:${CALLERID(num)})}) exten => _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr) exten =>

Hi, I'm currently trying to connect an Aastra 57i to our Asterisk Server. The strange thing is, that altough I have definitely entered the correct IP address of the server, the phone doesn't even attempt to register. Here is the configuration file (local.cfg) of the phone: firmware md5: dee6e938b469e217a87138076f47fe41 boot count: 1 tone set: Germany language 1: German time server1:

Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan --

Hi, I spent some hours working my way through the WIKI and a number of other documentations, but after all, three questions are still left: 1. I'm using a Fritz!Card with the i4l driver - no problem at all, my Grandstream BT 100 rings when I diall a regular phone number. Is there any need for me to configure the zapata.conf for the ISDN card to get a special asterisk feature or is this

I have a small Samba 2.2.1a plus Win2k workgroup. Samba is running on Redhat 7.2. There is a networked laserjet 5000 using a static IP which I can print to successfuly from the linux box, and from the win2k workstations via the queue on the linux box. The weird thing is that, on the win2k worksations, if I double click on the printer icon the status of the printer is reported as "Access

Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten => 83086921,1,Answer exten => 83086921,2,Dial(SIP/stefan,5,r) exten => 83086921,3,VoiceMail,u111 exten => 83086921,4,Hangup exten => 83086921,103,VoiceMail,b111 exten => 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 =>

Hello, we are curently benchmarking an asterisk system 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files Apr 19 14:20:51 WARNING[4045]: acl.c:306 ast_ouraddrfor: Cannot create socket Apr 19

Hi, is anyone on the list using the Siemens Openstage phones together with asterisk? If yes, is it possible to use the programmable keys of these phones together with Asterisk? Thanks for any hints, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93

Hi, I'm using an Aastra 57i together with Asterisk 1.4.13. The 57i is configured for call pickup as recommended by Aastra. When the LED flashes, I press the corresponding button and the display tells me "Call not possible". In the CLI is see the follwoing output. Why isn't the call transfered to user2 and what does "No target channel found for 31 mean"? User2 can

Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten => 83086921,1,Answer exten => 83086921,2,Dial(SIP/stefan,5,r) exten => 83086921,3,VoiceMail,u111 exten => 83086921,4,Hangup exten => 83086921,103,VoiceMail,b111 exten => 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 =>

Paul wrote > >;Pause/unpause Queue >exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)}) >exten => 424,2,Playback(unavailable) >exten => 424,3,Hangup >exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)}) >exten => 425,2,Playback(available) >exten => 425,3,Hangup > Following your suggestion and a number of postings and articles I have

Hello, I have read the articles on echo cancellation (http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a solution to my problem. We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M PCI (current driver from EICON) and some SNOM 300/360. There are few clients where we recognize echoes on both sides when we call them via ISDN. With some of these clients we

I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks