similar to: Same action to multiple numbers

Hi all! I am working with asterisk and trying to get it work in DHCP network, where asterisk gets a DHCP address as well as other computers (IP-phones). So far, I've have got asterisk working with static IP address where phones are getting their IP from DHCP server. Is it possible at all to phones to find asterisk server if it gets random IP address from the DHCP server also? I mean, is there

Hi! I'm trying to start asterisk at boottime. Since SuSe has no rc.local like in Redhat linux, I need asterisk starting script to /etc/init.d/rc3.d -directory (I assume it is like that if i want automated asterisk startup). Do you have any experience how this is implemented in SuSe, and if you have some useful script for starting asterisk, I would be very, i mean VERY pleased? Thank you all

Hi! I have just installed Redhat 9 and Asterisk to my computer, and now i have problems with my non-zaptel Card, I don't know how to set it up since all instructions are for digium's hardware. I have searched from the Internet for hours now, can you help me to understand all this HFC-s thing and how it is related to CAPI, ISDN4Linux, bristuff and so on. I have to say that I am not so

Hey! My asterisk is working properly so far with all automatic functions. Now I want to direct incoming calls to operator, i mean some person who answers to the incoming calls and redirect them to the person caller wants. What I have so far searched from the voip-info.org and other sites, Ihave not found any example configuration how to do it. First i thought that I will implement this just using

Hi! I'm trying to build gnugk with asterisk. Asterisk is working well with chan_h323 built with needed PWlib v.1.5.2 and open H.323 v.1.12.2. But gnugk' s installing instructions says that I need latest PWlib(1.17.1) and openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's compiling fails and I get error 1. Do you have any working solutions with asterisk and

I''m trying to setup some mildly complex associations for a project we''re working on and can''t seem to find much documentation on n-way has_many :through associations. I have the following models: Person, PhysicalAddress, EmailAddress, PhoneNumber. Each person can have multiple PhysicalAddresses, EmailAddresses, and PhoneNumbers, and multiple people can share the same

Hi! I have installed two hfc-s cards to handle my pstn calls. I use mISDN with capi, so capi.conf is edited. I have tested both separate and cards are working well. But they are not working together. It seems that when i set up settings for the other card: ;capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=50 incomingmsn=* controller=1

Is it possible to change error messages for Validation helpers? I am writing an app against a existing database (so no control over column names), but when there is validation error (e.g. with validate_presence_of) I would like to customize the field name. For example for telephone whose field name is PhoneNumber I would like to chnage it to "Telephone Number cannot be empty" rather

Hi, I?m looking for settings to configure net2phone carrier in my asterisk. I found this configurations, but it?s not work. I don?t known if this configuration is for voice line or voice access account. Anybody can help me, with other configuration? Thanks. ---- *sip.conf* [general] useragent = X-Lite release 1103m register => PHONENUMBER:PASSWORD@sip.net2phone.com [net2phone] type = peer

Hi! I have a working asterisk-setup with four sip-clients. Everything works great but when the users call someone the phonenumber shows up on the receiving ends callerid-display. To correct this my provider told me to send #31# before the phonenumber, tried this with: Dial(SIP/#31#${EXTEN}@provider) but my asterisk tells me that it isn't a valid extension. The INVITE looks fine,

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

Hi all, I've been trying to use BroadVoice as a SIP service provider. They don't officially support * but are helpful when it comes to answering questions for setup parameters. They claim they have no firewalls or access lists that need to be set up so I can get access to their servers. However, something's still not quite right when I use the parameters. It looks like our Asterisk

I am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",

I have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT)

Hello! My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on Fedora Core 4. I am making automatic call-out campaign with this setup on 4 PRI. The scripts for this: ==================================================================== caller php script write this to outgoung folder: fwrite($outfile,"Channel: Zap/g1/$phonenumber\n"); fwrite($outfile,"MaxRetries:

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

Hi folks! Our company are going to buy an E1 line with Euro ISDN and 30 lines (channels). This is how it will be configured: 3 Lines, of the total of 30, is going to be for the company phones, and share one phonenumber (eg. 555-12340). 1 Line will be dedicated to a specific unique phonenumber (Fax) (eg. 555-54321). The rest of the lines/channels (26) will be used by (by, not for) our customers,

hello list, i cant create an isolated network with the virt-manager. installed version virt-manager 3.0.0 installed version libvirt 6.2.0 output in error-message: Error creating virtual network: internal error: Failed to apply firewall rules /sbin/iptables -w --table filter --insert LIBVIRT_INP --in-interface virbr3 --protocol tcp --destination-port 67 --jump ACCEPT: iptables: No

Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --------------------------------------------------------------- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco