Re: [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.
Hello Experts, Could anybody pl help resolve my query? Thanks Regards, Subbaiah Nachiappan From: NACHIAPPAN, SUBBAIAH (SUBBAIAH) Sent: Tuesday, October 28, 2014 6:04 PM To: 'asterisk-users@lists.digium.com' Subject: RE: Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface. Hello Folks, Forgot to mention the software Versions which I am using: Asterisk: 1.8 Free PBX: 2.11 Asterisk NOW: 5.211.65 Thanks Regards, Subbaiah Nachiappan From: NACHIAPPAN, SUBBAIAH (SUBBAIAH) Sent: Tuesday, October 28, 2014 5:52 PM To: 'asterisk-users@lists.digium.com' Subject: Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface. Hello, I am new to Asterisk forum :). I have a requirement of terminating 3G Mobile originated calls (coming through 3G-MSC) to EPBX Phones via Asterisk PBX. Setup: Mobile Mobile Switching Center ( 3G)-SIP interface---Asterisk PBX---SIP Phone. I wanted to know if I require SIP licenses to integrate 3G MSC with my Asterisk server. I know there is a wealth of information in wiki link, but I am unable to locate the required configuration document which will help me in integrating MSC with Asterisk EPBX via SIP interface. Thanks in Advance!!! Thanks Regards, Subbaiah Nachiappan -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.commailto:asterisk-users-requ...@lists.digium.com Sent: Tuesday, October 28, 2014 5:41 PM To: NACHIAPPAN, SUBBAIAH (SUBBAIAH) Subject: Welcome to the asterisk-users mailing list -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.
On 29/10/14 12:59 pm, A J Stiles wrote: Imagine what would have happened to the human race if Ugg the Caveman decided not to share the secret of making fire with everyone freely, but instead went around demanding shiny beads with menaces from anyone who just wanted to keep themselves warm . That's the best analogy I've heard in favour of open development for a long time, and something that non-techs can understand. I thank you sir :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.
Hello, I am new to Asterisk forum :). I have a requirement of terminating 3G Mobile originated calls (coming through 3G-MSC) to EPBX Phones via Asterisk PBX. Setup: Mobile Mobile Switching Center ( 3G)-SIP interface---Asterisk PBX---SIP Phone. I wanted to know if I require SIP licenses to integrate 3G MSC with my Asterisk server. I know there is a wealth of information in wiki link, but I am unable to locate the required configuration document which will help me in integrating MSC with Asterisk EPBX via SIP interface. Thanks in Advance!!! Thanks Regards, Subbaiah Nachiappan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com Sent: Tuesday, October 28, 2014 5:41 PM To: NACHIAPPAN, SUBBAIAH (SUBBAIAH) Subject: Welcome to the asterisk-users mailing list -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.
Hello Folks, Forgot to mention the software Versions which I am using: Asterisk: 1.8 Free PBX: 2.11 Asterisk NOW: 5.211.65 Thanks Regards, Subbaiah Nachiappan From: NACHIAPPAN, SUBBAIAH (SUBBAIAH) Sent: Tuesday, October 28, 2014 5:52 PM To: 'asterisk-users@lists.digium.com' Subject: Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface. Hello, I am new to Asterisk forum :). I have a requirement of terminating 3G Mobile originated calls (coming through 3G-MSC) to EPBX Phones via Asterisk PBX. Setup: Mobile Mobile Switching Center ( 3G)-SIP interface---Asterisk PBX---SIP Phone. I wanted to know if I require SIP licenses to integrate 3G MSC with my Asterisk server. I know there is a wealth of information in wiki link, but I am unable to locate the required configuration document which will help me in integrating MSC with Asterisk EPBX via SIP interface. Thanks in Advance!!! Thanks Regards, Subbaiah Nachiappan -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.commailto:asterisk-users-requ...@lists.digium.com Sent: Tuesday, October 28, 2014 5:41 PM To: NACHIAPPAN, SUBBAIAH (SUBBAIAH) Subject: Welcome to the asterisk-users mailing list -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query list of defined channel variables via AMI
Is there a way to list the names of the channel variables that are currently defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar needs the name of the variable to get. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query list of defined channel variables via AMI
Check DumpChan http://www.voip-info.org/wiki/view/Asterisk+cmd+DumpChan http://wikiasterisk.com/index.php/Aplicaciones_B%C3%A1sicas#Aplicaci.C3.B3n_DumpChan Regards, Rafael Rincón IP-COM, Inc Senior Network Engineer rrin...@ipcomnetwork.com 3100 SW 145th Ave. Suite 410 Miramar, FL 33027 +1 (305) 477 2902 Miami x 111 +1 (877) 55 IPCOM US Toll Free x 111 +52 (55) 3692 4266 Mexico City x 111 + 57 (1) 742-3408 Bogota, Colombia x 111 CONFIDENTIALITY NOTICE The information contained in this email is intended only for the individual or entity to whom it is addressed. It may contain confidential and privileged information and if you are not an intended recipient, you must not copy, distribute or take any action in reliance upon it. If you believe you have received the email in error or doubt the authenticity of email apparently from this source, please notify the sender. You should then destroy and delete the message from your computer. On Dec 3, 2012, at 10:17 AM, Alex Villací s Lasso wrote: Is there a way to list the names of the channel variables that are currently defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar needs the name of the variable to get. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query result is array of elements, how to iterate over it ??
Hey list,
suppose I have several dates in a database-table, where these dates are
marked as 'set' or 'not set'.
If I do something like :
SELECT ID FROM my_table WHERE client=clientID AND set=yes
and this query results in several rows and thus several ID's like 2 5 7
11 13 14 17...
How can I iterate over these values ??
Something like :
for each item in ARRAY {
SELECT asterisk_syntax FROM other_table WHERE dateID=$item
}
This query would result in something like *|*|7|may.
How to iterate over an array of results ???
Jonas.
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Re: [asterisk-users] Query about tdm410 cards
On Wed, 10 Jun 2009, Alex Samad wrote: Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I know this mighs sound odd, but do you really need the +12V connection? You only need it if you have analogue phones plugged in and not exchange lines.. I know - this is obvious and you probably do have analogue phones plugged in, but I'm just checking!!! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about tdm410 cards
On Thu, Jun 11, 2009 at 09:02:37AM +0100, Gordon Henderson wrote: On Wed, 10 Jun 2009, Alex Samad wrote: Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I know this mighs sound odd, but do you really need the +12V connection? You only need it if you have analogue phones plugged in and not exchange lines.. I have 2 fxs + 1fxo so I know - this is obvious and you probably do have analogue phones plugged in, but I'm just checking!!! we all miss the obvious at some time Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- If we are going to save a generation of young people, our children must know they will face bad consequences for criminal behavior. Sadly, too many youths are not getting that message. Our juvenile justice system must say to our children: We love you, but we are going to hold you accountable for your actions. - George W. Bush 01/01/2000 2000 Bush campaign literature signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about tdm410 cards
At 02:01 PM 6/10/2009, you wrote: http://www.cyberguys.com/product-search/?keyword=molex doesn't look like it, really need a 90 degree plug and I am in OZ not usa so postage is going to kill me I'd buy a standard one, pull the pins, cut off the wire end of the plug, put it back in bend the pins over and insulate it with a bit of hot melt or heatshrink. Probably as good as anything you'll buy. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about tdm410 cards
On Thu, Jun 11, 2009 at 11:14:47AM -0700, Ira wrote: At 02:01 PM 6/10/2009, you wrote: http://www.cyberguys.com/product-search/?keyword=molex doesn't look like it, really need a 90 degree plug and I am in OZ not usa so postage is going to kill me I'd buy a standard one, pull the pins, cut off the wire end of the plug, put it back in bend the pins over and insulate it with a bit of hot melt or heatshrink. Probably as good as anything you'll buy. I have soldered to the back of the board, the molex pins go all the way through the pcb Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I'm not available for comment.. signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query about tdm410 cards
Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. I have read up about a PWR2400b and it seems to use 2wire pin, I am guessing to connect to P8 just behind the molex connector on the tdm410. can any one here confirm this, or have any info on the pwr2400b - ie how it connects to the cards. The web site is a bit devoid of the information and all the photo's are not clear. this would make my life rather simple, I have 12V + GND to supply the card - seems like people have done this with a TDM400, unfortunately the 410 is longer Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about tdm410 cards
On Wed, Jun 10, 2009 at 7:17 AM, Alex Samada...@samad.com.au wrote: Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. You can probably find the extension cable or connector you need here http://www.cyberguys.com/product-search/?keyword=molex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about tdm410 cards
On Wed, Jun 10, 2009 at 08:44:22AM -0400, David Backeberg wrote: On Wed, Jun 10, 2009 at 7:17 AM, Alex Samada...@samad.com.au wrote: Hi recently bought a soekris net5501 and a tdm410 to place in there. I am having some issues attaching 12V power to the card via the molex card - basically the box for the motherboard is too small. You can probably find the extension cable or connector you need here http://www.cyberguys.com/product-search/?keyword=molex doesn't look like it, really need a 90 degree plug and I am in OZ not usa so postage is going to kill me thanks -- Why is it that all of the instruments seeking intelligent life in the universe are pointed away from Earth? signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about tdm410 cards
Alex Samad wrote: I have read up about a PWR2400b and it seems to use 2wire pin, I am guessing to connect to P8 just behind the molex connector on the tdm410. can any one here confirm this, or have any info on the pwr2400b - ie how it connects to the cards. The web site is a bit devoid of the information and all the photo's are not clear. No, the PWR2400B includes a PCI bracket with cables that connect to the Molex connectors on the cards. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about tdm410 cards
On Wed, Jun 10, 2009 at 05:49:22PM -0500, Kevin P. Fleming wrote: Alex Samad wrote: I have read up about a PWR2400b and it seems to use 2wire pin, I am guessing to connect to P8 just behind the molex connector on the tdm410. can any one here confirm this, or have any info on the pwr2400b - ie how it connects to the cards. The web site is a bit devoid of the information and all the photo's are not clear. No, the PWR2400B includes a PCI bracket with cables that connect to the Molex connectors on the cards. oh well I have soldering iron, seems like a few people have soldered to the connectors underneath Alex -- My plan reduces the national debt, and fast. So fast, in fact, that economists worry that we're going to run out of debt to retire. - George W. Bush 02/24/2001 radio address signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query About Asterisk 1.6.0.1 Dialog Event Package.
Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and x-lite2 phone(X-lite) the configurations are as below for sip.conf. My Scenario. I am using asterisk-1.6.0.1, I want to know that does it support for the Dialog Event Package (As per the rfc 4235)). If yes then what config files I have to change for the same. Example of my scenario Here App is My Application. x-lite1 is X-lite Phone. x-lite2 is X-lite phone. 1) Register a subscriber App, and then subscribe it to (for Dialog Event Package) x-lite1 through asterisk. 2)After subscribing I(App) receive NOTIFY from asterisk. 3) Now establish call between x-lite1 and x-lite2 both x-lite phones. 4) After the call is established from x-lite1 to x-lite2, then an NOTIFY should be sent to App (for the change in the dialog event of x-lite1), further App should get notified for any dialog change by the x-lite1. My problem. Here App is subscribed to X-lite1 through asterisk for the dialog event package, and X-lite1 calls X-lite2, here the dialog events of X-lite1 should be notified to App. But I am getting 404 Not Found for the Subscribe message, which I send for Subscription to x-lite1 through asterisk. Below are the config files. Is there any other way to solve this problem. Any help is appreciated. Thank you in advance. Sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 192.168.1.243 ; x = Asterisk server IP address disallow=all ;allow = ulaw ; Allow all codecs ;allow = alaw context = from-sip ; Send SIP callers that we don't know about here canreinvite=no directrtpsetup=yes nat=no ;subscribecontext= localextensions ;default allowsubscribe=yes ; Disable support for subscriptions. (Default is yes) [App] type=friend username=App ;regexten=1234 ; When they register, create extension 1234 ;secret=password host=dynamic context=from-sip mailbox=App disallow=all allow = alaw ;canreinvite=no ;directrtpsetup=yes subscribecontext=internal ;localextensions ;default allowsubscribe=yes [X-lite1] type=friend username=X-lite1 ;secret=password host=dynamic context=from-sip mailbox=X-lite1 disallow=all allow = alaw ;canreinvite=no ;directrtpsetup=yes subscribecontext=internal ;localextensions ;default allowsubscribe=yes [X-lite2] type=friend username= X-lite2 ;secret=password host=dynamic context=from-sip mailbox= X-lite2 disallow=all allow = alaw ;canreinvite=no ;directrtpsetup=yes subscribecontext=internal ; localextensions ;default allowsubscribe=yes The below configuration is for extension.conf. extension.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here. [from-sip] exten = App,1,Dial(SIP/App,20) exten = App,2,Hangup exten = X-lite1,1,Dial(SIP/X-lite1,20) exten = X-lite1,2,Hangup exten = X-lite2,1,Dial(SIP/X-lite2,20) exten = X-lite2,2,Hangup Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at http://www.techmahindra.com/Disclaimer.html externally and http://tim.techmahindra.com/Disclaimer.html internally within Tech Mahindra. Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at a href=http://www.techmahindra.com/Disclaimer.html;http://www.techmahindra.com/Disclaimer.html/a externally and a href=http://tim.techmahindra.com/Disclaimer.html;http://tim.techmahindra.com/Disclaimer.html/a internally within Tech Mahindra. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query about Call Recording with Asterisk / Freeswitch in Cisco IPCC deployment
Hello, One of our client company is providing hosted contact center solutions with Cisco IPCC. To keep the Call Recording cost at low, they are planning to use Asterisk / Free Switch. Can anyone integrate Cisco IPCC with Asterisk for call recording ? Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query about Bluetooth Head phone
Hello All, Can anybody suggest bluetooth head phone which can be used to place calls with eyebeam or any other soft phone. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about Bluetooth Head phone
Ya i had some months ago. it works fine. what you need to know else... JehanZaib Younis Date: Wed, 19 Mar 2008 14:04:35 +0500From: [EMAIL PROTECTED]: asterisk-users@lists.digium.com; [EMAIL PROTECTED]: Query about Bluetooth Head phone Hello All, Can anybody suggest bluetooth head phone which can be used to place calls with eyebeam or any other soft phone. Regards, -- Kashif NaeemBusiness Development ManagerHadi Telecomwww.haditelecom.comCell: +92 (0)345 4226006Office: +92 (0)42 5692766Email: [EMAIL PROTECTED]: [EMAIL PROTECTED]: [EMAIL PROTECTED]: kashif.naeem302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, I am running asterisk PBX ( digium TE120P card configured) on one system. It is connected to E1 card running application on the other system. After establishing sync between two card, I am able to place call from sip phone to E1 card running application. I want to pass the callerid, when calling from sip phone to E1 card running application. Which all configuration files is to be changed in the asterisk. I am doing the following changes in extensions.conf exten=115,1,Dial(ZAP/g1/115,20) So, extension 115 is received at other end as callerid. Is it correct. Can any body help in how to configure for callerid with digium card. thanks and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
You can set the caller-id in many different ways but the easiest in by setting it in the sip.conf profile for the extension. So you can just add a line like this to your sip.conf under the extension: callerid=Your Name 5554441212 Hope this helps.. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com On 8/8/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am running asterisk PBX ( digium TE120P card configured) on one system. It is connected to E1 card running application on the other system. After establishing sync between two card, I am able to place call from sip phone to E1 card running application. I want to pass the callerid, when calling from sip phone to E1 card running application. Which all configuration files is to be changed in the asterisk. I am doing the following changes in extensions.conf exten=115,1,Dial(ZAP/g1/115,20) So, extension 115 is received at other end as callerid. Is it correct. Can any body help in how to configure for callerid with digium card. thanks and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
Hi Sanchal, 115 in your case is just DIALLED NUMBER and it will be searched by you E1 trunk. If you want change your CALLERID, you would insert one default or would insert one to each user. the command is the same sendt by Todd: callerid=Your Name 5554441212 but you can work with function callerid and set up it in the same extensions. more informations about it, you have in http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid all the best and good luck, Thiago Maluf. 2007/8/8, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hi, I am running asterisk PBX ( digium TE120P card configured) on one system. It is connected to E1 card running application on the other system. After establishing sync between two card, I am able to place call from sip phone to E1 card running application. I want to pass the callerid, when calling from sip phone to E1 card running application. Which all configuration files is to be changed in the asterisk. I am doing the following changes in extensions.conf exten=115,1,Dial(ZAP/g1/115,20) So, extension 115 is received at other end as callerid. Is it correct. Can any body help in how to configure for callerid with digium card. thanks and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi , I am trying to dial in from two sip phones on one end, through digium card to E1 card running application on another end. with following configuration /etc/asterisk/zapata.conf group=1 context=default euroisdn=EuroISDN signalling= pri_net context=incoming channel=1-15,17-31 /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 /etc/asterisk/sip.conf [phone1] type=friend host=192.168.1.67 dtmfmode=rfc2833 context=sip port=5060 nat=yes [phone2] type=friend host=192.168.1.53 dtmfmode=rfc2833 context=sip port=5060 nat=yes /etc/asterisk/extension.conf [sip] exten=112,1,Dial(SIP/phone2,20,tr) ; Dialing from sip phone1 at one system (192.168.1.67)through ; through soft switch to sip Phone2 (192.168.1.53) running at ; at other system having IP 192.168.1.53 exten=113,1,Dial(ZAP/1,16) ; Dialing from sip phone1 at one system (192.168.1.67) through ; asterisk PBX having digium card to other E1 ; card running application exten=115,1,Dial(ZAP/1,16) [incoming] exten=114,1,Dial(SIP/phone1,20,tr) ; Making call from E1 card running application ; to soft switch through digium card and ; diverting to sip phone1 rinning on system ; 192.168.1.67 I am able to dial from phone1 to E1 card running application successfully but when I dial from phone2 to Ei card running application it gives error message. app_dial.c:1076dial_exec_full:unable to create channel of type ZAP(cause 0 unknown) Everyone is busy/conjusted at this time (1:0/0/1) auto fall through channel 'SIP/192.168.1.53/081c63b8' Status is CHANUNAVAILABLE. Can anybody help me to solve this problem. thanks regards Sanchal Singh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, I am able to dial through asterisk PBX having TE120P card to E1 card running application. Communication was established successfully Now, I want to do the reverse way out. I am using the following configurations 1)zaptel.conf span=1,1,0,ccs,hdb3,crc4 defaultzone=us bchan=1-15,17-31 dchan=16 2)zapata.conf group=1 signalling=pri_net switchtype=euroisdn context=incoming channel=1-15,17-31 What configuration changes is to be done for landing of call to asterisk PBX when dialled from E1 card running application. I was trying to dial out from E1 card running application with extension number 114 and added the following lines in extensions.conf of asterisk configuration files exten=114,1,Dial(SIP/Phone1,20,tr) but asterisk debugging console is giving the error message -- Extension '114' in context 'channelbank' from '' does notexist. Rejecting call on channel 0/1, span 1 Can anybody tell me how to handle the configuration files for extension number to be called from E1 card running application. Thanx and regards, sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
On 26 Jul 2007 17:25:30 +0530, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channelanyway! Can anybody tell me how to overcome this error. Sanchal: If you will refer to my message of two days ago it explains exactly how to fix the issue. Best regards, Andrew ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, Do the following steps are required while configuring D-channel 1) In zconfig.h file of zaptel package uncomment #define CONFIG_ZAPATA_NET make sethdlc-new make install 2) modprobe wcte12xp ztcfg 3) sethdlc hdlc0 cisco Step 3 is giving error hdlc0: Unable to set Cisco HDLC protocol information: No such device Can anybody tell, how to overcome this error. Thanx and regards, sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
I expect you mean /etc/asterisk/zapata.conf, not zaptel.conf. You say DIGIUM card is connected through cable to another end. What is at the other end? If it is a PBX that thinks it's connected to the PSTN, then it is a cpe (customer premise equipment) and you would want to specify signalling=pri_net to indicate that you're taking the role of the network. (I have responded directly to you, as well as to the list, because I have been experiencing delays in receiving messages from the list. You may already have received solutions to your issue that I won't see for another day or more.) --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, July 26, 2007 6:56 AM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [asterisk-users] Query Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to asterisk PBX ( through DIGIUM card ) the following error messages is coming on console mode of asterisk Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channelanyway! Can anybody tell me how to overcome this error. Thanx and Regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
On Thu, Jul 26, 2007 at 05:25:30PM +0530, [EMAIL PROTECTED] wrote: Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf /etc/asterisk/zapata.conf Right? group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to asterisk PBX ( through DIGIUM card ) the following error messages is coming on console mode of asterisk Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! What is on the other side? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to asterisk PBX ( through DIGIUM card ) the following error messages is coming on console mode of asterisk Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channelanyway! Can anybody tell me how to overcome this error. Thanx and Regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to asterisk PBX ( through DIGIUM card ) the following error messages is coming on console mode of asterisk (The OTHER END CONNECTED to DIGIUM is E1 CARD RUNNING APPLICATION) Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! NOTE- The OTHER END CONNECTED to DIGIUM is E1 CARD RUNNING APPLICATION Can anybody tell me how to overcome this error. Thanx and Regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
Hi, It looks like your configuration file zapata.conf syntax is wrong. Have a look in the sample files how to set it up correctly, and if you are still having troubles, paste your zapata.conf here. Cheers, Dimitri [EMAIL PROTECTED] wrote: Hi, I have put Digium TE120P card in PCI slot. So, lspci command gives the information in followimg format. 02:0a.0 Ethernet controller: Unknown device d161:0120 (rev 11) Following modules are running when seen through lspci wcte11xp 22304 - ztdynamic 9804 - ztdummy 3468 - ip_conntrack_irc6640 - ip_conntrack_ftp7312 - ipt_state 1864 - iptable_mangle 2696 - ipt_REJECT 5160 - ipt_LOG 6280 - ipt_multiport 2376 - ip_conntrack 47524 - iptable_filter 2856 - ipt_limit 2280 - ip_tables 18168 - wcte12xp 44352 - zaptel180036 - but on running asterisk -vvvgc it stops by printing the following errrors '###' at line 41 of /etc/asterisk/zapata.conf Jun 26 15:19:38 WARNING[1691]: config.c:497 process_text_line: Unknown directive '###' at line 43 of /etc/asterisk/zapata.conf Jun 26 15:19:38 NOTICE[1691]: chan_zap.c:10388 setup_zap: Ignoring unknown keyword 'group' in trunkgroups Jun 26 15:19:38 WARNING[1691]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Jun 26 15:19:38 ERROR[1691]: chan_zap.c:7050 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jun 26 15:19:38 ERROR[1691]: chan_zap.c:10484 setup_zap: Unable to register channel '1-31' Jun 26 15:19:38 WARNING[1691]: loader.c:415 __load_resource: chan_zap.so: load_module failed, returning -1 Jun 26 15:19:38 WARNING[1691]: loader.c:555 load_modules: Loading module chan_zap.so failed! What is the problem actually can anybody tell me. Thanx and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
For this you have to make entry in sip.conf. it will be better if you use host=dynamic in both the phones in sip.conf and what is the IP you are putting in phones which are on your PC. Also check that your both sip phones which are on PC, are sending requestr to asterisk server or not. Kesh. [EMAIL PROTECTED] wrote: Hi, I am trying to establish call through sip phone between two PC connected to linux box on which asterisk server is running 1st PC is having IP Adress : 192.168.1.149 2nd PC is having IP Adress : 192.168.1.53 I am trying to dial from 1st PC to 2nd PC through asterisk server The problem is 1st PC is calling directly to 2nd PC not through asterisk server I am doing the following additions in configuration files 1) sip.conf [general] context=sip bindport=5060 bindaddr=0.0.0.0 [phone1] type=friend host=192.168.1.149 port=5060 nat=yes dtmfmode=rfc2833 context=sip [phone2] type=friend host=192.168.1.53 port=5060 nat=yes dtmfmode=rfc2833 context=sip 2) extensions.conf exten = 11,1,Dial(SIP/phone2,20,tr) Now, I am calling from sip phone1 by name [EMAIL PROTECTED] It is not being called through asterisk server running on linux m/c. as no packet dumping us taking palce. As, I am running sip debub no messages are seen on screen. What additional routing informations are to be added to sip.conf, inorder to make it work . Thanx and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, I am trying to establish call through sip phone between two PC connected to linux box on which asterisk server is running 1st PC is having IP Adress : 192.168.1.149 2nd PC is having IP Adress : 192.168.1.53 I am trying to dial from 1st PC to 2nd PC through asterisk server The problem is 1st PC is calling directly to 2nd PC not through asterisk server I am doing the following additions in configuration files 1) sip.conf [general] context=sip bindport=5060 bindaddr=0.0.0.0 [phone1] type=friend host=192.168.1.149 port=5060 nat=yes dtmfmode=rfc2833 context=sip [phone2] type=friend host=192.168.1.53 port=5060 nat=yes dtmfmode=rfc2833 context=sip 2) extensions.conf exten = 11,1,Dial(SIP/phone2,20,tr) Now, I am calling from sip phone1 by name [EMAIL PROTECTED] It is not being called through asterisk server running on linux m/c. as no packet dumping us taking palce. As, I am running sip debub no messages are seen on screen. What additional routing informations are to be added to sip.conf, inorder to make it work . Thanx and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, I am trying to establish call through sip phone between two PC connected to linux box on which asterisk server is running 1st PC is having IP Adress : 192.168.1.149 2nd PC is having IP Adress : 192.168.1.53 Now, I am tying to dial from 1st PC to 2nd PC I am trying to dial from 1st PC to 2nd PC through asterisk server The problem is 1st PC is calling directly to 2nd PC not through asterisk server I am doing the following additions in configuration files 1) sip.conf [general] context=sip bindport=5060 bindaddr=0.0.0.0 [phone1] type=friend host=192.168.1.149 port=5060 nat=yes dtmfmode=rfc2833 context=sip [phone2] type=friend host=192.168.1.53 port=5060 nat=yes dtmfmode=rfc2833 context=sip 2) extensions.conf exten = 11,1,Dial(SIP/phone2,20,tr) Now, I am calling from sip phone1 by name [EMAIL PROTECTED] It is not being called through asterisk server running on linux m/c. It is calling directly. As, I am running sip debub but no packet dumping is taking place. Can anybody will tell me the error I am doing. Thanx and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
On 6/28/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am trying to establish call through sip phone between two PC connected to linux box on which asterisk server is running 1st PC is having IP Adress : 192.168.1.149 2nd PC is having IP Adress : 192.168.1.53 Now, I am tying to dial from 1st PC to 2nd PC I am trying to dial from 1st PC to 2nd PC through asterisk server ... 2) extensions.conf exten = 11,1,Dial(SIP/phone2,20,tr) Now, I am calling from sip phone1 by name [EMAIL PROTECTED] Why? Shouldn't you just pick up Phone1 and dial 11? If you dial it by the IP address, why would it go through Asterisk? It is not being called through asterisk server running on linux m/c. It is calling directly. As, I am running sip debub but no packet dumping is taking place. Can anybody will tell me the error I am doing. I am going to assume that the typo is in the above paragraph, and you really mean sip debug. If not, that's another problem. Thanx and regards sanchal Hope that helps, David ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
El Thu, Jun 28 de 2007 a las 20:18 +0530, [EMAIL PROTECTED] comentaba:
Hi,
I am trying to establish call through sip phone between two PC
connected to linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
Now, I am tying to dial from 1st PC to 2nd PC
I am trying to dial from 1st PC to 2nd PC through asterisk server
The problem is 1st PC is calling directly to 2nd PC not through
asterisk server
I am doing the following additions in configuration files
1) sip.conf
[general]
context=sip
bindport=5060
bindaddr=0.0.0.0
[phone1]
type=friend
host=192.168.1.149
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
[phone2]
type=friend
host=192.168.1.53
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
2) extensions.conf
exten = 11,1,Dial(SIP/phone2,20,tr)
Now, I am calling from sip phone1 by name [EMAIL PROTECTED]
I guess thats why the phones are talking directly: [EMAIL PROTECTED]
Either call extension '11' from phone1 or add a extension named 'phone2' to
extensions.conf and call that extension ('phone2') without the ip address.
Make sure your softphones are correctly configured: sip proxy address (*
address), username, etc.
Btw, both devices in sip.conf are declared as 'friend'.. thus you must specify
a secret (and optionally a username):
[phone2]
type=friend
username=phone2
secret=qwerty
host=192.168.1.53
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
It is not being called through asterisk server running on linux m/c. It
is calling directly. As, I am running sip debub but no packet dumping is
taking place. Can anybody will tell me the error I am doing.
Thanx and regards
sanchal
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Re: [asterisk-users] Query
I am not sure what exactly you wish to achieve. Just a basic SIP--to--SIP call
or ?
I am not much into the configs, but ya I can tell you that you can try using
FreePBX or Trixbox kind of setup to write ur Asterisk config file, rather then
u editing them, as it has macros, context etc... which is too high to me. But
the browser interface help a lot understanding the config files later once
configured via FreePBX.
FreePBX -- Its a tool(software which is wrapper over asterisk which gives a
web based interface to manage configure ur asterisk configuration files with
easy understanding.
tixbox-- Its a kind of Asterisk solution which is combination of
asterisk+freepbx+linux+crm tools etc.. for quick Asterisk deployment.
I am not sure whether u know all these if yes, hen excuse me.. but ur mail
sounded u might need this info needed.
[EMAIL PROTECTED] wrote:
Hi,
I am trying to establish call through sip phone between two PC connected to
linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
Now, I am tying to dial from 1st PC to 2nd PC
I am trying to dial from 1st PC to 2nd PC through asterisk server
The problem is 1st PC is calling directly to 2nd PC not through asterisk server
I am doing the following additions in configuration files
1) sip.conf
[general]
context=sip
bindport=5060
bindaddr=0.0.0.0
[phone1]
type=friend
host=192.168.1.149
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
[phone2]
type=friend
host=192.168.1.53
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
2) extensions.conf
exten = 11,1,Dial(SIP/phone2,20,tr)
Now, I am calling from sip phone1 by name [EMAIL PROTECTED]
It is not being called through asterisk server running on linux m/c. It is
calling directly. As, I am running sip debub but no packet dumping is taking
place. Can anybody will tell me the error I am doing.
Thanx and regards
sanchal
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[asterisk-users] Query
Hi, I have put Digium TE120P card in PCI slot. So, lspci command gives the information in followimg format. 02:0a.0 Ethernet controller: Unknown device d161:0120 (rev 11) Following modules are running when seen through lspci wcte11xp 22304 - ztdynamic 9804 - ztdummy 3468 - ip_conntrack_irc6640 - ip_conntrack_ftp7312 - ipt_state 1864 - iptable_mangle 2696 - ipt_REJECT 5160 - ipt_LOG 6280 - ipt_multiport 2376 - ip_conntrack 47524 - iptable_filter 2856 - ipt_limit 2280 - ip_tables 18168 - wcte12xp 44352 - zaptel180036 - but on running asterisk -vvvgc it stops by printing the following errrors '###' at line 41 of /etc/asterisk/zapata.conf Jun 26 15:19:38 WARNING[1691]: config.c:497 process_text_line: Unknown directive '###' at line 43 of /etc/asterisk/zapata.conf Jun 26 15:19:38 NOTICE[1691]: chan_zap.c:10388 setup_zap: Ignoring unknown keyword 'group' in trunkgroups Jun 26 15:19:38 WARNING[1691]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Jun 26 15:19:38 ERROR[1691]: chan_zap.c:7050 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jun 26 15:19:38 ERROR[1691]: chan_zap.c:10484 setup_zap: Unable to register channel '1-31' Jun 26 15:19:38 WARNING[1691]: loader.c:415 __load_resource: chan_zap.so: load_module failed, returning -1 Jun 26 15:19:38 WARNING[1691]: loader.c:555 load_modules: Loading module chan_zap.so failed! What is the problem actually can anybody tell me. Thanx and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, Can any body tell me (i) Does digium TE-120P card can be installed on Redhat linux 9i (2.4.20-8) kernel (ii) It is written in documentation that TE120P card be installed only above 2.6.xx . So, which is the best suited one for it( 2.6.15 or 2.6.18 os some other release) (iii) Redhat 9i (2.4.20-8) is installed on my system. I downloaded 2.6.18 kernel. compiled and installed it. After booting through the new one, when I give lsmiod command, it gives the following error lsmod: QM_MODULES: Function not implemented Unable to load iptables module I tried the following way of kernel trap 1. Download the latest version of module-init-tools. 2. ./configure --prefix=/ make make instal 3. Now translate your old /etc/modules.conf into /etc/modprobe.conf with the ./generate-modprobe.conf script that comes with module-init-tools: ./generate-modprobe.conf /etc/modprobe.conf It worked for once. But everyday morning same problem of lsmod comes. I could not find out the way to remove this error of lsmod. Can anybody tell me the way to sort it out. Thanx and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
On Mon, Jun 25, 2007 at 12:10:04PM +0530, [EMAIL PROTECTED] wrote: Hi, Can any body tell me (i) Does digium TE-120P card can be installed on Redhat linux 9i (2.4.20-8) kernel Why do you keep starting a new thread and not bother following up to answers in existing threads? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very serious problem of modutils and iptable. Can anybody help me out. Thanx and Regards sanchal singh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of modutils and iptable. Can anybody help me out of this. Thanx and Regards sanchal singh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
On Fri, Jun 22, 2007 at 03:20:07PM +0530, [EMAIL PROTECTED] wrote: Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very serious problem of modutils and iptable. What problems, exactly? While it should be possible to install the card on such a system, is there any good reason you keep using such an old and unmaintained OS? If you're used to working with the RedHat way, why not try Centos (or buy RHEL)? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
On 6/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of modutils and iptable. Can anybody help me out of this. Thanx and Regards sanchal singh Either contact digium support or post the problem ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
ram wrote: On 6/22/07, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of modutils and iptable. Can anybody help me out of this. Thanx and Regards sanchal singh Either contact digium support or post the problem ram Time for CentOS. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
The best person to check with is Digium support. They have support matrix for
Kernel hardware on which ur card will perform.
Please check the compatibility matrix. Should work fine with
http://www.digium.com/en/supportcenter/documentation/viewdocs/TE120P
Digium support. 256-428-6000
[EMAIL PROTECTED] wrote: Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card on
redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of
modutils and iptable.
Can anybody help me out of this.
Thanx and Regards
sanchal singh
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[asterisk-users] Query regarding connecting PABX with Application server
Dear all, We are connecting the PABX with Application server.What we are trying is that when a nbr 1800 (this is not registered in PABX) is dialled the pabx should route the call to Application server .The PABX should also have intelligence to route the call by itself for its registered clients.For this scenario to work please guide us what are the files we need to change and other necessary details.If possible please also provide us the configuration that needs to be set up in XLITE sip soft phone for this service. Kindly do the needful. Thanks and regards, S.Ravi___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about DTMF generate
Hello Lee, Thanks a lot thats right but in i hearing tone when i click on buton but it not take asterisk as a DTMF generate code so voice mail not identified. thats problem . if u knw then reply me. Regards, gaur On 5/18/07, Lee Jenkins [EMAIL PROTECTED] wrote: gaurang sheladiya wrote: Hello every body, kindly i make one phone which is in java applet and there is no generate any DTMF signal at client side only beep tones is hearing but not generate DTMF at the back end side. so plz if anyone know that DTMF generation proccess then plz reply me. Thank you. After reading your post again, I thought maybe this would be of use more than other links I provided: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+indications.conf -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query about DTMF generate
Hello every body, kindly i make one phone which is in java applet and there is no generate any DTMF signal at client side only beep tones is hearing but not generate DTMF at the back end side. so plz if anyone know that DTMF generation proccess then plz reply me. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about DTMF generate
gaurang sheladiya wrote: Hello every body, kindly i make one phone which is in java applet and there is no generate any DTMF signal at client side only beep tones is hearing but not generate DTMF at the back end side. so plz if anyone know that DTMF generation proccess then plz reply me. Thank you. Asterisk cmd SendDTMF: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SendDTMF General Google Search of www.voip-info.org: http://www.google.com/custom?tk=b44a2457968317bad5a3domains=www.voip-info.org Have a great day. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query about DTMF generate
gaurang sheladiya wrote: Hello every body, kindly i make one phone which is in java applet and there is no generate any DTMF signal at client side only beep tones is hearing but not generate DTMF at the back end side. so plz if anyone know that DTMF generation proccess then plz reply me. Thank you. After reading your post again, I thought maybe this would be of use more than other links I provided: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+indications.conf -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query Failed because: Incorrect information in file: './asterisk/sip.frm'
Hi, I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today I encountered this error. Now, I have no acces to any information in mysql realtime, so nothing work now ! [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten = 'h' AND context = ' interne' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten = 'h' AND context = 'interne ' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND conte xt = 'interne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context = 'i nterne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/sip.frm 'ast ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query Failed because: Incorrect information in file: './asterisk/sip.frm'
Its a problem in your database. something might have corrupted...be prepared to load a backup Gregory Duchatelet wrote: Hi, I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today I encountered this error. Now, I have no acces to any information in mysql realtime, so nothing work now ! [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten = 'h' AND context = ' interne' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten = 'h' AND context = 'interne ' AND priority = '1' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND conte xt = 'interne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context = 'i nterne' AND priority = '1' ORDER BY exten [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/extensi ons.frm' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query: SELECT * FROM sip WHERE name = '129.200.1.51' [Jan 24 10:32:40] DEBUG[31026] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect information in file: './asterisk/sip.frm 'ast ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query regarding Pulse Dialing at 20 pps
Hi, I have a query regarding pulse dialing at 20 pps. An Analog Phone is directly connected to the FXS port of Asterisk PBX. When the analog phone pulse-dials at 20 pps, the pulse digits were not decoded correctly by Asterisk. For e.g. when the user dials a 2, Asterisk decodes the pulse digit as 1. Does anyone know how this problem could be solved ? Thank you. regards, Kwang Mien ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query on Call Parking
Hi, I understand that 700 is the default extension to initiate a Call Park. Does anyone know of a way to configure Asterisk such that it has more than one park extension for e.g. parkexten = 700,800,900 regards, Kwang Mien ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query on MWI
Hi users; i am new in the mailing list and asterisk user . i have to implement METHOD 3 of the link (http://www.voip-info.org/wiki/view/Asterisk+at+largeview_comment_id=11963) i have question that is: Q:when lets i have getting a NOTIFY message and my phone changes the tone to a MWI tone now if i restart the Telephone adapter i loose the tone so how do i fix this? Thanks and Regards Tanzeel __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query ,NEED help regarding MWI
Hi Users; i have to implement MWI scenario like this: IPphone,ATAopenserAsterisk my users are registered at openser and voicemail box is configured at asterisk. MWI is send by ASTERISK to OPENSER and then OPENSER to IPPHONE OR ATA. My query is this; Q:let say i got a NOTIFY message from openser to IPPHONE/ATA and my phone changes the tone to a MWI tone, Now if the Telephone adapter got reset by any reason , the MWI tone gets lost ,so HOW CAN I FIX THIS PROBLEM? I mean how openser know that phone got disconnected so that it again send NOTIFY message to IPPHONE or is there any way to send NOTIFY message peridically after a define time stamp. Hope someone would help me Thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query on Call Forward Feature codes for SIP users..
This is what I do:
[cf]exten = _*72XXX,1,DBput(CF/${CALLERIDNUM}=${CALLERIDNUM:-10:3}${EXTEN:3})exten = _*72XXX,2,Answerexten = _*72XXX,3,Playback(call-fwd-unconditional)exten = _*72XXX,4,Playback(is-set-to)
exten = _*72XXX,5,SayDigits(${EXTEN:3})exten = _*72XXX,6,Playback(vm-goodbye)exten = _*72XXX,7,hangupexten = _*72X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:3})exten = _*72X.,2,Answer
exten = _*72X.,3,Playback(call-fwd-unconditional)exten = _*72X.,4,Playback(is-set-to)exten = _*72X.,5,SayDigits(${EXTEN:3})exten = _*72X.,6,Playback(vm-goodbye)
exten = _*72X.,7,hangupexten = *73,1,DBdel(CF/${CALLERIDNUM})exten = *73,2,Answerexten = *73,3,Playback(call-fwd-cancelled)exten = *73,4,wait(.5)exten = *73,5,playback(vm-goodbye)
exten = *73,6,hangup
*72+ number will activate call forwarding... *73 will deactivate call forwarding.
You then just add a DBget in your inbound dialplan to see if CF key exists in the database.
On 9/7/06, A C Sathish-a22713 [EMAIL PROTECTED] wrote:
All,Could any one help me in configuring the feature codes for Callforward feature in asterisk..?
Howto configure the feature code *XX for activation /deactivation ofcall forward for SIP users ?Would appreciate , if somebody can help me more in detail .Thanks Regards,-Sathish
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[asterisk-users] Query on Call Forward Feature codes for SIP users..
All, Could any one help me in configuring the feature codes for Call forward feature in asterisk..? How to configure the feature code *XX for activation /deactivation of call forward for SIP users ? Would appreciate , if somebody can help me more in detail . Thanks Regards, -Sathish ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query about Call Detail Record in Asterisk
Hi, My testbed is as follows: sipphone -- Asterisk PBX 1 -- Asterisk PBX 2 -- PSTN -- Analog Phone I understand that one of the fields in the CDR (Call Detail Record) is the Answer field which is the time when call is answered. Is it right that : a) the Answer field of the CDR at Asterisk PBX 1 shows the time when Asterisk PBX 2 answers the call from Asterisk PBX 1 ? b) the answer field of the CDR at Asterisk PBX 2 shows the time when the Analog Phone answers the call from Asterisk PBX 2 ? regards, Kwang Mien ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Query
Hi, Can anybody tell me Does Asterisk has a TAPI Interface sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] query
Hi, Can anybody tell me that does asterisk have TAPI interface sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] query
[EMAIL PROTECTED] wrote: Hi, Can anybody tell me that does asterisk have TAPI interface sanchal No, if you're a windows user, there is asttapi which uses the management interface though. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Query: IAXModem
Hi, I am in a problem. Can anybody help me out. I am trying to establish connection using hyperterminal through IAXsoft modem using asterisk PBX. I have done the following settings in the configuraion files of asterisk. 1) iax.conf file: [iaxmodem] type=friend ;username=iaxmodem ;secret=n19d19 host=dynamic qualify=yes ;trunk=yes ;context=in-fax disallow=all allow=ulaw allow=alaw allow=gsm [iaxmodem1] type=friend ;type=peer ;username=iaxmodem ;secret=n19d19 host=dynamic qualify=yes ;trunk=yes ;context=in-fax ;not required allow=ulaw allow=alaw allow=gsm 2)extensions.conf file exten = 12,1,Dial(IAX2/iaxmodem1) //for dialing to modem2 with extension number 12 through modem1 using hyperterminal 3) created ttyIAX0 and ttyIAX1 file in path /etc/iaxmodem ttyIAX0 file: device /dev/ttyIAX0 owner uucp:uucp mode 660 port 4571 refresh 100 server 127.0.0.1 peername iaxmodem secret password #Cidname John0 #Cidnumber 8005551212 codec slinear ttyIAX1 file: device /dev/ttyIAX1 owner sanchal:uucp mode 660 port 4572 refresh 300 server 127.0.0.1 peername iaxmodem1 #secret password cidname John2 cidnumber 8005551231 codec slinear 4)Now using hyper terminal send atdt12 from one side it sends ring to other side . On replying ATA from other side, it sends connect but not in accordance with class1 format. Client other end at+fclass=1 -- -- at+fclass=1 OK ---- OK atdt12 -- -- ring connect -- -- ATA -- connect NOCARRIER -- --ERROR Can anybody give me the guidelines how to proceed further to transfer a file after establishing a successful connection. Regards sanchal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] query about Three way calling
Hi all I am a new user of asterisk and want to implement three way calling but not getting any information about how to configure asterisk conf file, Dial Plan etc.. whether a Digium card is essential or not? Is Three way calling is same as MeetMe or separate feature in asterisk? Can You please send me some Document or link where i can get all relevant information. thanks in advance himanshu The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s)and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender or [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] query about Three way calling
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, February 01, 2006 7:14 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] query about Three way calling Hi all I am a new user of asterisk and want to implement three way calling but not getting any information about how to configure asterisk conf file, Dial Plan etc.. whether a Digium card is essential or not? Is Three way calling is same as MeetMe or separate feature in asterisk? Can You please send me some Document or link where i can get all relevant information. thanks in advance himanshu If you only need 3 way calls, most IP phones / ATAs will handle this for you without having to configure anything in * using their conference feature. Keep it simple. Meetme is most typically used for conference bridging were each participant calls the conference bridge directly, this is different than one user originating 2 calls. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] query about cdr configuration
hi friends ! can anybody tell me something about cdr configuration. actually i want to confirm about the minimum requiremnts. is it possible to configure it with mysql server and myodbc anly or unixodbc is also required? in case unixodbc is also requied than help me to send some download links that already have worked well. thanks Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] query about cdr configuration
Hello Deepak, yes, you can use mysql. the packages are in asterisk-addons. there is a very good wiki page on the subject here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql hope this helps, yair On Apr 6, 2005 7:33 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: hi friends ! can anybody tell me something about cdr configuration. actually i want to confirm about the minimum requiremnts. is it possible to configure it with mysql server and myodbc anly or unixodbc is also required? in case unixodbc is also requied than help me to send some download links that already have worked well. thanks Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
