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  • DNS resolve without depending on router for asterisk system

    - by john
    Hello, Basically I have a Debian box running asterisk assigned an IP via DHCP with host-name XXX. My windows browser can resolve the host-name but if I use host-name in X-Lite or my SPA922 phone it fails to resolve. Is there any way of getting this to work without depending on the router or assigning a static IP (request is to make it portable). I was thinking zero-conf but am unsure (box has limited HDD too). Any help is most appreciated.

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  • Asterisk with new functions

    - by MhdAboAhmad
    hi all , please need help for an asterisk I created a write func odbc list records files in sql table: [R] dsn=connector write=INSERT INTO ast_records (filename,caller,callee,dtime) VALUES ('${ARG1}','${ARG2}','${ARG3}','${ARG4}') prefix=M and set it in dialplan : exten = _0X.,n,Set( M_R(${MIXMONITOR_FILENAME}\,${CUSER}\,${EXTEN}\,${DTIME})= ) when I excute it I get an error : ast_func_write: M_R Function not registered:

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  • How can I do a fastAgi to use with Asterisk in Perl

    - by Edwin Jaws
    Hi! I am trying to do an IVR using FAstAGI to say information from my database to my clients. I done AGI doing this but I need now run this from another server, windows server, but I dont know how can I do this. I checked the Asterisk::fastagi module but it is so confused and I dont understand anything. I did a few AGI perl scripts without problem but now I get. If somebody can give me an example to simple script to put me on rigth direction to do that will be so appreciated. Just I need one exmaple and to know if all AGI normal commands from AGI like stream_file are available in fastagi Thks In Advance Edwin Jaws

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  • Asterisk: Originate API - Which card to use to detect busy/ringing/answer event for FXO

    - by spkhaira
    I want to use Originate API of Asterisk to place an outbound call on a FXO channel, for testing purpose I am using X100P card and, as expected, card is not able to detect if the number is busy/ringing or when it is answered. I want to know which card should I use so that I can get such basic events ... I am not really interested in detailed call progress analysis for answering machine or live voice. I just need basic busy/ringing and answer events and maybe a dis-connect event. Thanks.

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  • Asterisk : SpeechBackground application.

    - by abinila
    Hai everyone, I have used the SpeechBackground application in asterisk. I used the version 1.6.0.6. I have a entry like, ;;SpeechCreate exten => s,1,SpeechCreate() exten => s,2,SpeechActivateGrammar(yesno) exten => s,3,SpeechStart() exten => s,4,SpeechBackground(demo-instruct) exten => s,5,SpeechDeactivateGrammar(yesno) I don't know which file I meed to give in SpeechBackground application. Please give me any idea. I have given the sound file from /sounds directory. If I call to 's' the call will be immediately released.I didn't get any audio sound. Please any one help me...

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  • Asterisk doesn't start properly at system startup. DNS lookup fails.

    - by leiflundgren
    When I start my Ubuntu system it attempts two DNS lookups. One to find out what my internet-routers external ip is. And one to find the IP of my PSTN-SIP-provider. Both fails. [Apr 7 22:14:54] WARNING[1675] chan_sip.c: Invalid address for externhost keyword: sip.mydomain.com ... [Apr 7 22:14:54] WARNING[1675] acl.c: Unable to lookup 'sip.myprovider.com' And since the DNS fails it cannot register properly a cannot make outgoing or incoming calls. If I later, after bootup, restart asterisk everything works excelent. Any idea how I should setup things so that either: Delay Asterisk startup so that DNS is up and healthy first. Somehow get Asterisk to re-try the DNS thing later. Regards Leif

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  • Asterisk doesn't start properly at system startup. DNS lookup fails.

    - by leiflundgren
    When I start my Ubuntu system it attempts two DNS lookups. One to find out what my internet-routers external ip is. And one to find the IP of my PSTN-SIP-provider. Both fails. [Apr 7 22:14:54] WARNING[1675] chan_sip.c: Invalid address for externhost keyword: sip.mydomain.com ... [Apr 7 22:14:54] WARNING[1675] acl.c: Unable to lookup 'sip.myprovider.com' And since the DNS fails it cannot register properly a cannot make outgoing or incoming calls. If I later, after bootup, restart asterisk everything works excelent. Any idea how I should setup things so that either: Delay Asterisk startup so that DNS is up and healthy first. Somehow get Asterisk to re-try the DNS thing later. Regards Leif

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  • Automatically reconnect to ODBC sources?

    - by stefan.at.wpf
    I am using Asterisk 1.8.10.1 and a MySQL database connected via ODBC to store CDRs. When my MySQL database isn't available when Asterisk starts or has an outage while Asterisk is running, I would expect Asterisk to retry to connect to the database, but this doesn't happen! Anyone knows where I can enable some kidn of automatic reconnect to databases in Asterisk? My res_odbc.conf looks like this: [asterisk] enabled => yes dsn => asterisk-connector username => user password => pass pre-connect => yes pooling => no limit => 1 idlecheck => 1 negative_connection_cache => 1

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  • Asterisk TDM Out Channel Not Recording

    - by Benny
    I am trying to use the monitor command to record a TDM extension, but only the in chnnnel is being recorded. The out channel is 44 bytes and obviously no audio within. However, when monitoring a SIP or IAX phone, no problems exist. Is there some configuration I'm missing for distinguishing between TDM and SIP/IAX for recording? Thanks in advance!

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  • Problems with IRQs when connecting two digium card in and asterisk box

    - by JorgeO
    I have two Digium Wildcard TDM800P with 8 FXO ports each. When I connect both at the same time IRQ misses start showing up making my computer unresponsive and unusable. One card works fine but I need all 16 FXO ports to work to receive calls from my Telco. Is there a way for the cards to communicate with each other so they don't generate as many interrupts. Or a way to tweak Linux to dedicate separate IRQ's for each card. I have tried disabling Audio, ACPI and USB ports. Still too many IRQ misses.

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  • How to choose an open source, Asterisk friendly firewall?

    - by Lucas
    I'm in pain. We are moving to a SIP based VOIP system and for whatever reason, we could not get our hosted Asterisk solution to work with our Sonicwall. Our VOIP provider gave up and is recommending an open source vendor, pfSense. A little background: We have about 30 users in our network. We use a few IPSec VPN connections for remote networks. I would like, but don't need, application layer filtering. We're active internet users, so properly traffic shaping is probably a concern. How can I tell if an open source firewall will handle VOIP setup smoothly with a hosted Asterisk system?

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  • CRMIT Solution´s CRM++ Asterisk Telephony Connector Achieves Oracle Validated Integration with Oracle Sales Cloud

    - by Richard Lefebvre
    To achieve Oracle Validated Integration, Oracle partners are required to meet a stringent set of requirements that are based on the needs and priorities of the customers. Based on a Telephony Application Programming Interface (TAPI) framework the CRM++ Asterisk Telephony Connector integrates the Asterisk telephony solutions with Oracle® Sales Cloud. "The CRM++ Asterisk Telephony Connector for Oracle® Sales Cloud showcases CRMIT Solutions focus and commitment to extend the Customer Experience (CX) expertise to our existing and potential customers," said Vinod Reddy, Founder & CEO, CRMIT Solutions. "Oracle® Validated Integration applies a rigorous technical review and test process," said Kevin O’Brien, senior director, ISV and SaaS Strategy, Oracle®. "Achieving Oracle® Validated Integration through Oracle® PartnerNetwork gives our customers confidence that the CRM++ Asterisk Telephony Connector for Oracle® Sales Cloud has been validated and that the products work together as designed. This helps reduce deployment risk and improves the user experience for our joint customers." CRM++ is a suite of native Customer Experience solutions for Oracle® CRM On Demand, Oracle® Sales Cloud and Oracle® RightNow Cloud Service. With over 3000+ users the CRM++ framework helps extend the Customer Experience (CX) and the power of Customer Relations Management features including Email WorkBench, Self Service Portal, Mobile CRM, Social CRM and Computer Telephony Integration.. About CRMIT Solutions CRMIT Solutions is a pioneer in delivering SaaS-based customer experience (CX) consulting and solutions. With more than 200 certified customer relationship management (CRM) consultants and more than 175 successful CRM deployments globally, CRMIT Solutions offers a range of CRM++ applications for accelerated deployments including various rapid implementation and migration utilities for Oracle® Sales Cloud, Oracle® CRM On Demand, Oracle® Eloqua, Oracle® Social Relationship Management and Oracle® RightNow Cloud Service. About Oracle Validated Integration Oracle Validated Integration, available through the Oracle PartnerNetwork (OPN), gives customers confidence that the integration of complementary partner software products with Oracle Applications and specific Oracle Fusion Middleware solutions have been validated, and the products work together as designed. This can help customers reduce risk, improve system implementation cycles, and provide for smoother upgrades and simpler maintenance. Oracle Validated Integration applies a rigorous technical process to review partner integrations. Partners who have successfully completed the program are authorized to use the “Oracle Validated Integration” logo. For more information, please visit Oracle.com at http://www.oracle.com/us/partnerships/solutions/index.html.

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  • ASP.NET MVC–How to show asterisk after required field label

    - by DigiMortal
    Usually we have some required fields on our forms and it would be nice if ASP.NET MVC views can detect those fields automatically and display nice red asterisk after field label. As this functionality is not built in I built my own solution based on data annotations. In this posting I will show you how to show red asterisk after label of required fields. Here are the main information sources I used when working out my own solution: How can I modify LabelFor to display an asterisk on required fields? (stackoverflow) ASP.NET MVC – Display visual hints for the required fields in your model (Radu Enuca) Although my code was first written for completely different situation I needed it later and I modified it to work with models that use data annotations. If data member of model has Required attribute set then asterisk is rendered after field. If Required attribute is missing then there will be no asterisk. Here’s my code. You can take just LabelForRequired() methods and paste them to your own HTML extension class. public static class HtmlExtensions {     [SuppressMessage("Microsoft.Design", "CA1006:DoNotNestGenericTypesInMemberSignatures", Justification = "This is an appropriate nesting of generic types")]     public static MvcHtmlString LabelForRequired<TModel, TValue>(this HtmlHelper<TModel> html, Expression<Func<TModel, TValue>> expression, string labelText = "")     {         return LabelHelper(html,             ModelMetadata.FromLambdaExpression(expression, html.ViewData),             ExpressionHelper.GetExpressionText(expression), labelText);     }       private static MvcHtmlString LabelHelper(HtmlHelper html,         ModelMetadata metadata, string htmlFieldName, string labelText)     {         if (string.IsNullOrEmpty(labelText))         {             labelText = metadata.DisplayName ?? metadata.PropertyName ?? htmlFieldName.Split('.').Last();         }           if (string.IsNullOrEmpty(labelText))         {             return MvcHtmlString.Empty;         }           bool isRequired = false;           if (metadata.ContainerType != null)         {             isRequired = metadata.ContainerType.GetProperty(metadata.PropertyName)                             .GetCustomAttributes(typeof(RequiredAttribute), false)                             .Length == 1;         }           TagBuilder tag = new TagBuilder("label");         tag.Attributes.Add(             "for",             TagBuilder.CreateSanitizedId(                 html.ViewContext.ViewData.TemplateInfo.GetFullHtmlFieldName(htmlFieldName)             )         );           if (isRequired)             tag.Attributes.Add("class", "label-required");           tag.SetInnerText(labelText);           var output = tag.ToString(TagRenderMode.Normal);             if (isRequired)         {             var asteriskTag = new TagBuilder("span");             asteriskTag.Attributes.Add("class", "required");             asteriskTag.SetInnerText("*");             output += asteriskTag.ToString(TagRenderMode.Normal);         }         return MvcHtmlString.Create(output);     } } And here’s how to use LabelForRequired extension method in your view: <div class="field">     @Html.LabelForRequired(m => m.Name)     @Html.TextBoxFor(m => m.Name)     @Html.ValidationMessageFor(m => m.Name) </div> After playing with CSS style called .required my example form looks like this: These red asterisks are not part of original view mark-up. LabelForRequired method detected that these properties have Required attribute set and rendered out asterisks after field names. NB! By default asterisks are not red. You have to define CSS class called “required” to modify how asterisk looks like and how it is positioned.

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  • Build Your Own PBX With Asterisk and Linux

    Setting up your own Asterisk installation isn't for the faint of heart, but the savings you can reap from combining the powerful, open source PBX with Linux are worth the effort. Here's a quick guide to getting your own Asterisk install up and running.

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  • How to display incoming trunk name at called device end with Asterisk?

    - by netmano
    We have an Asterisk with FreePBX, and using Grandstream and Panasonic VoIP phones. Now when an incoming call rings on the phones only displays "Line 1" (as it is configured at account 1 on phone) and the caller number. We would need to see the trunk name where the call comes. Please, suggest how can be achieved. I was wondering about rewriting the callerID to extend by the trunk name (like "LP - +12345").

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  • Is it possible to monitor an Asterisk ConfBridge from a URL / Browser?

    - by Lorin S.
    I have an Asterisk server set up with minimal configuration, including the following confbridge definition / extension: *confbridge.conf* [testbridge] type=bridge video_mode=follow_talker max_members=20 mixing_interval=10 internal_sample_rate=auto record_conference=yes *extension.conf* exten => 6100,1,Answer() same => n,Set(CONFBRIDGE(user,admin)=yes) same => n,Set(CONFBRIDGE(user,marked)=yes) same => n,ConfBridge("Ad-hoc",testbridge,default_user,sample_user_menu) Is it possible to monitor the video / audio of the conference without joining via a client?

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  • Asterisk server firewall script allows 2-way audio from incoming calls, but not on outgoing?

    - by cappie
    I'm running an Asterisk PBX on a virtual machine directly connected to the Internet and I really want to prevent script kiddies, l33t h4x0rz and actual hackers access to my server. The basic way I protect my calling-bill now is by using 32 character passwords, but I would much rather have a way to protect The firewall script I'm currently using is stated below, however, without the established connection firewall rule (mentioned rule #1), I cannot receive incoming audio from the target during outgoing calls: #!/bin/bash # first, clean up! iptables -F iptables -X iptables -t nat -F iptables -t nat -X iptables -t mangle -F iptables -t mangle -X iptables -P INPUT ACCEPT iptables -P FORWARD DROP # we're not a router iptables -P OUTPUT ACCEPT # don't allow invalid connections iptables -A INPUT -m state --state INVALID -j DROP # always allow connections that are already set up (MENTIONED RULE #1) iptables -A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT # always accept ICMP iptables -A INPUT -p icmp -j ACCEPT # always accept traffic on these ports #iptables -A INPUT -p tcp --dport 80 -j ACCEPT iptables -A INPUT -p tcp --dport 22 -j ACCEPT # always allow DNS traffic iptables -A INPUT -p udp --sport 53 -j ACCEPT iptables -A OUTPUT -p udp --dport 53 -j ACCEPT # allow return traffic to the PBX iptables -A INPUT -p udp -m udp --dport 50000:65536 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT iptables -A INPUT -p udp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -p tcp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -m multiport -p udp --dports 10000:20000 iptables -A INPUT -m multiport -p tcp --dports 10000:20000 # IP addresses of the office iptables -A INPUT -s 95.XXX.XXX.XXX/32 -j ACCEPT # accept everything from the trunk IP's iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT # accept everything on localhost iptables -A INPUT -i lo -j ACCEPT # accept all outgoing traffic iptables -A OUTPUT -j ACCEPT # DROP everything else #iptables -A INPUT -j DROP I would like to know what firewall rule I'm missing for this all to work.. There is so little documentation on which ports (incoming and outgoing) asterisk actually needs.. (return ports included). Are there any firewall/iptables specialists here that see major problems with this firewall script? It's so frustrating not being able to find a simple firewall solution that enabled me to have a PBX running somewhere on the Internet which is firewalled in such a way that it can ONLY allows connections from and to the office, the DNS servers and the trunk(s) (and only support SSH (port 22) and ICMP traffic for the outside world). Hopefully, using this question, we can solve this problem once and for all.

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  • Do something by operator dial specified number in Asterisk?

    - by Rev
    I want to make ability for Asterisk phone Operators to able do something like this: While operator talking to caller, if Operator dial specified number like 244 (or something like that but not Sip-Userid's), do something (like play sound for caller or etc) for that call. So, Is this possible? Is need to change dialplan? ¦¦¦¦¦ I found this. in first paragraph it's say someething like: if operator dial exten go voiceMail.

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