search for: phonenumb

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 from. # U CISCO_IP:503...

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 from....

...ossible 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 than "Phonenumber cannot be empty". Is that possible? --Jeet -- Posted via http://www.ruby-forum.com/.

Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there

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

Hello, Can anyone suggest us how to got the phonenumber of the caller. In the environment variable, I just see the ip of the gateway. Environment: 'agi_callerid' is 'XXX.xxx.XXX.xx' Should I do some changes in the GW conf, or is it just not possible when we got a call from PSTN ?!? Thx in advance, Ares

...figure 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 host = sip.net2phone.com username = PHONENUMBER secret = PASSWORD fromuser = PHONENUMBER fromdomain = net2phone.com context = incoming insecure = very canreinvite = no *extensions.conf* [outgoing] exten => _9NXXNXXXXXX,1,Dial(SIP/net2phone/$...

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, '#31#<phonenumber>@provide...

Hi Guys, I am trying to make a web form where a person is allowed to put in $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller ID. There are a few problems that I am facing with Asterisk AMI Originate command. The reason why I want to use the darn AMI Originate is because I am sending calls to mobile phones and I want to have some accountability and to know...

...ry to call our assigned BroadVoice number, I immediately get a BroadVoice message saying the number is busy. I can provide the sip debug output but it basically shows that BroadVoice appears to be not communicating with inbound or outbound requests. Here are the entries: sip.conf register => phonenumber:password@sip.broadvoice.com:5060/phonenumber [broadvoice] type=friend username=phonenumber secret=password host=proxy.broadvoice.com dtmfmode=inband disallow=all allow=ulaw allow=alaw extensions.conf ; calls via BroadVoice exten => _6NXXXXXX,1,Dial,SIP/1925${EXTEN:1}@broadvoice exten =>...

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", "Zap/g1/[phonenumber]|60") in new stack -- Called g1/[phonenumber] -- Zap/1-1 answered SIP/sipphone-9eb0 And then I get silence. The phone doesn't ring on the other end. I ha...

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 PhysicalAddress, EmailAddress, or PhoneNumber. I need to track the types of associations (i.e. home, work, cell, etc) for each, so habtm definitely won''t cut it. D...

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

...cht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von bladerunner Gesendet: Freitag, 6. Mai 2005 14:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] CAPI on ptp with variable length digits in phonenumber hi again, just ignore my mentioning of the sirrix-cards, just realised you have a PRI, i overread it and thought you had a BRI. so i think your last hope is a zaptel-card. regards, Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin: > Hi! > > we have a german PtP PRI connection...

Hello! I have an Asterisk@home instalation with 7 users working OK, and I'ld like to implement either a -- Web dial feature, where the user would fill one form field with a phone number and a connection would be created between his extention and the entered number. OR -- Dial using an URI (callto:xxxxx link in a web page), having AstTapi installed and configured in all workstations.

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 'phonenumber@sip.broadvoice.com'" If I do keewi*CLI> sip show peers Name/username Host Dyn Nat ACL Mask P...

Hello Everyone, I am making a simple index.php file which will allow a web user to enter his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged. Following is the index.php and the contents of extensions_custom.conf. When I submit the form nothing happens. I don't even see Manager Connected msg. Your input will be much appreciated. I am thinking I have some syntax problem....

...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) Using INVITE request as basis request - 41597440-0-320116780 at carrierIP Found peer 'phonenumber' for 'phonenumber' from carrierIP:5060 <--- Reliably Transmitting (NAT) to carrierIP:5060 ---> SIP/2.0 401 Unauthorized -------- Good call --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT) Using INVITE request as basis request - 41604639-0-321360830 at carrier...

...ith 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($outfile,"MaxRetries: 0\n"); fwrite($outfile,"RetryTime: 5\n"); fwrite($outfile,"WaitTime: 20\n"); fwrite($outfile,"Context: 0100q\n"); fwrite($outfile,"Callerid: $dbid\n"); fwrite($outfile,"Extension: $phonenumber\n");...

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into