similar to: Aleatori PSTN number with SIP.

Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =>usuario1:pass1@<public_ip_2> In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=<public_ip_1> dtmfmode=inband Logs in * are the followings In * one logs: Sip

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

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

Thks for your answer, here is an exemple of what i do with the errors in french... > tmp [1] 200 150 245 125 134 345 320 450 678 > beta18 Erreur : Objet "beta18" not found //NORMAL just to show it > eta [1] 500 > func1<-function(beta18) dweibull(tmp[1],beta18,eta) > func1<-func1(beta18) * function(beta18) dweibull(tmp[2],beta18,eta) Erreur dans dweibull(tmp[1],

Folks, I have a Merlin Legend R7 V10.0 with a 2 100D cards. I have 1 card in slot 4 going to CenturyTel, and the card in slot 10 going to a flip cable to a TE110P card in a Asterisk 1.6.x box. I have routing setup on the Merlin to send a block of numbers to the Asterisk. Currently the PSTN can dial the Asterisk Extensions. The Asterisk can dial Merlin Extensions. The Merlin can Dial Asterisk

Hi list, I'm in central-europe and signed yesterday a broadvoice account. My Asterisk box is CVS 2005-08-25. Problem I face is: "Failed to authenticate on REGISTER to 'phonenumber@sip.broadvoice.com' (Tries 2)" then "Registration for 'phonenumber@sip.broadvoice.com' timed out" and finaly "Giving up forever to register

Hello, LARTC mailing readers, I hope u can help with this mysterious issue i''m having with my linux box acting as a router. Scenario: Linux running 2.6.8.1 /w julians patches Latest iproute (iproute2-ss040702) 4 NICS ----------------- | x eth0 (63.43.x.x) network mask (255.255.240.0) | | x eth1 (63.43.x.x)

Hello, LARTC mailing readers, I hope u can help with this mysterious issue i''m having with my linux box acting as a router. Scenario: Linux running 2.6.8.1 /w julians patches /w support for multipath routing Latest iproute (iproute2-ss040702) 4 NICS ----------------- | x eth0 (63.43.x.x) network mask (255.255.240.0) | |

Hi everybody, Firstly I'm going to describe the scenario where I'm working. I use a E400P with Asterisk CVS-05/22/03-11:14:50, and I'm working with asterisk trow AGI scripts (Perl). The configuration of extension.conf is: exten =>_s,1,Answer exten =>_s,2,AGI,script.agi Inside the AGI script is call Dial application as follows: print "EXEC Dial

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 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,

I am trying to use the simple CID name management script on the wiki. http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names I can not see what is wrong. The values never get entered in the database. Here are the files: I have asterisk running as the user asterisk also. ---cid-store.php---- <HTML> <HEAD> <TITLE>Storing Asterisk CID data</TITLE> </HEAD>

Martin Joseph wrote: > > > For all of us using these devices, I have some good news. There is a > self installable firmware update available from Nokia here (requires > windows box to install): > > http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate > > This seems to radically improve the behavior of the SIP client on my > E60. It seems to have

Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!) asterisk -vvvvgc results after hanging up the pstn line in: -- Executing Hangup("SIP/1087997-d79f", "") in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation

I have two partials to deal with contact information. The Contact information uses single-table polymorphism. I want to be able to use save on Contact and set ''type'' manually, based on the type of form the user is filling in. This is saved as the value on a hidden field. class Contact belongs_to :person acts_as_list :scope => :person validates_presence_of :type

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:

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

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