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  • Asterisk + FreePBX + GoTalk. Inbound route not working.

    - by user289581
    I'm running asterisk 1.6.2.6 and freepbx-2.7.0 My trunk is configured as follows: Outgoing Settings Trunk name: GoTalk Peer Details: host=sip.gotalk.com username=09xxxxxx secret=YNxxxxxx type=peer fromuser=09xxxxxx fromdomain=sip.gotalk.com canreinvite=no insecure=very Incoming Settings User Context: 09xxxxx User Details: username=09xxxxx fromuser=09xxxxx type=peer secret=YNxxxxx insecure=very host=dynamic fromdomain=sip.gotalk.com context=from-pstn Register String: 09xxxxxx:[email protected]/09xxxxxx I have an inbound route called Incoming with DID 09xxxxxx diverted to local extension 200 When I do a sip trace and dial my telephone number 0741xxxxx I just get failure beeps. I never see any SIP traffic from GoTalk to my asterisk server trying to connect the call. Seems I'm not registering correctly for incoming calls because GoTalk aren't sending them to me. I am correct in using the GoTalk username 09xxxxxx as the DID, aren't I ? I've tried using my phone number but it makes no difference.

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  • Three Global Telecoms Soar With Siebel

    - by michael.seback
    Deutsche Telekom Group Selects Oracle's Siebel CRM to Underpin Next-Generation CRM Strategy The Deutsche Telekom Group (DTAG), one of the world's leading telecommunications companies, and a customer of Oracle since 2001, has invested in Oracle's Siebel CRM as the standard platform for its Next Generation CRM strategy; a move to lower the cost of managing its 120 million customers across its European businesses. Oracle's Siebel CRM is planned to be deployed in Germany and all of the company's European business within five years. "...Our Next-Generation strategy is a significant move to lower our operating costs and enhance customer service for all our European customers. Not only is Oracle underpinning this strategy, but is also shaping the way our company operates and sells to customers. We look forward to working with Oracle over the coming years as the technology is extended across Europe," said Dr. Steffen Roehn, CIO Deutsche Telekom AG... "The telecommunications industry is currently undergoing some major changes. As a result, companies like Deutsche Telekom are needing to be more intelligent about the way they use technology, particularly when it comes to customer service. Deutsche Telekom is a great example of how organisations can use CRM to not just improve services, but also drive more commercial opportunities through the ability to offer highly tailored offers, while the customer is engaged online or on the phone," said Steve Fearon, vice president CRM, EMEA Read more. Telecom Argentina S.A. Accelerates Time-to-Market for New Communications Products and Services Telecom Argentina S.A. offers basic telephone, urban landline, and national and international long-distance services...."With Oracle's Siebel CRM and Oracle Communication Billing and Revenue Management, we started a technological transformation that allows us to satisfy our critical business needs, such as improving customer service and quickly launching new phone and internet products and services." - Saba Gooley, Chief Information Officer, Wire Line and Internet Services, Telecom Argentina S.A.Read more. Türk Telekom Develops Benefits-Driven CRM Roadmap Türk Telekom Group provides integrated telecommunication services from public switched telephone network (PSTN) and global systems for mobile communications technology (GSM). to broadband internet...."Oracle Insight provided us with a structured deployment approach that makes sense for our business. It quantified the benefits of the CRM solution allowing us to engage with the relevant business owners; essential for a successful transformation program." - Paul Taylor, VP Commercial Transformation, Türk Telekom Read more.

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  • Commercial SIP Trunking in mainland China [closed]

    - by Patrick
    Is there any regulation preventing the use/sale of SIP trunks in mainland China? I've set up and used commercial-grade SIP trunks in places where previously we would have used ISDN T1/E1 connections. Here in Shanghai I'm looking for a similar service, and while E1 30B+D services are readily available, every telecoms company we speak with says that SIP trunking is not available in China with re-sellers of both China Telecom and China Unicom. But no one seems to know why. It seems logical to me that SIP trunks are cheaper to operate than ISDN services given that the first mile transit can be run over already-existing Internet infrastructure, and SIP signaling reduces the amount of configuration required by subscribers which is why it appeals to me. As such I've come to expect SIP services to be available in modern markets, and I've used them in quite a few countries. For example, one place I know it's not possible is in India. Government regulations in India make it illegal to provide PSTN service using VoIP. (Citations: 1, 2). However it seems this may be changing. Perhaps China has something similar.

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  • Asterisk terminating outbound call when picked up, sends 'BYE' message

    - by vo
    I'm running Asterisk 1.6.1.10 / FreePBX 2.5.2.2 and I've got an outbound trunk setup. Everything use to work fine until recently (perhaps due to upgrade to FC12 or other things I'm not sure). Anyway the setup does not appear to have issues registering and setting up the call, RTP packets go both ways and you can hear the ringing from the other side. However it appears that when the call is picked up or thereabouts, the incoming RTP packets cease. Upon closer inspection with Wireshark, there are these particular packets that seem to be the cause: trunk->asterisk SIP/SD Status: 200 OK, with session description asterisk->trunk SIP Request: ACK sip:<phone>@trunk:6889 asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 [..about a dozzen RTP packets in/outbound..] trunk->asterisk SIP Status: 200 OK, CSeq: 104 Bye [..outbound RTP continues, phone is silent..] Then the inbound RTP packets cease, however the asterisk logs dont show any activity at this point. The last entry reads 'SIP/ is answered SIP/'. Then when you hangup the extension, you get asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 trunk->asterisk SIP Status: 481 Call Leg/Transaction does not exist My trunk peer settings in FreePBX are: username=<user> fromuser=<user> canreinvite=no type=friend secret=<pass> qualify=no [qualify yes produces 401/forbidden messages] nat=yes insecure=very host=<sip trunk gateway> fromdomain=<sip trunk gateway> disallow=all context=from-pstn allow=ulaw dtmfmode=inband Under sip_general_custom.conf i have stunaddr=stun.xten.com externrefresh=120 localnet=192.168.1.1/255.255.255.0 nat=yes Whats causing Asterisk to prematurely end the call and still think the call is in progress? I have no idea where to look next.

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  • Looking for ballpark pricing on an affordable a Cisco VOIP solution for our office

    - by guytech
    We have about 8 incoming PSTN lines that are currently on an old and antiquated Nortel Meridian ICS system. This system has been giving us some grief. We're looking for a new VOIP solution. I've been looking at a Cisco solution and it does seem pricey but I'm sure effective. Unfortunately, we probably can't afford a Cisco Unified Communications 520 which seems to be the ideal solution. We have about 15 people who need an extension and voicemail. We really don't have any need for a fancy system just an auto attendant of some sort when people call us. It looks like we'll have to get an older router and an addon card for what we're looking for to get best value pricing. However, I don't know a a lot about Cisco voice products so I'm a bit lost as to what to get. The only thing I am sure on is the pricing on VOIP phones which we expect to be about ~$100-200. However, I'm not sure what pieces of VOIP infrastructure to get. Any advice? I am familiar with Asterisk but right now I'm looking on pricing concerning a Cisco solution.

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  • grep from a log file to get count

    - by subodh1989
    I have to get certain count from files. The grep statement i am using is like this : counter_pstn=0 completed_count_pstn=0 rec=0 for rec in `(grep "merged" update_completed*.log | awk '{print $1}' | sed 's/ //g' | cut -d':' -f2)` do if [ $counter_pstn -eq 0 ] then completed_count_pstn=$rec else completed_count_pstn=$(($completed_count_pstn+$rec)) fi counter_pstn=$(($counter_pstn+1)) done echo "Completed Orders PSTN Primary " $completed_count_pstn But the log file contains data in this format : 2500 rows merged. 2500 rows merged. 2500 rows merged. 2500 rows merged.2500 rows merged. 2500 rows merged. 2500 rows merged. As a result , it is missing out the count of one merge(eg on line 4 of output).How do i modify the grep or use another function to get the count. NOTE that the 2500 number maybe for different logs. So we have to use "rows merged" pattern to get the count. i have tried -o ,-w grep options,but it is not working. Expected output from above data: 17500 Actual output showing : 15000

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  • Lync Server 2010

    - by ManojDhobale
    Microsoft Lync Server 2010 communications software and its client software, such as Microsoft Lync 2010, enable your users to connect in new ways and to stay connected, regardless of their physical location. Lync 2010 and Lync Server 2010 bring together the different ways that people communicate in a single client interface, are deployed as a unified platform, and are administered through a single management infrastructure. Workload Description IM and presence Instant messaging (IM) and presence help your users find and communicate with one another efficiently and effectively. IM provides an instant messaging platform with conversation history, and supports public IM connectivity with users of public IM networks such as MSN/Windows Live, Yahoo!, and AOL. Presence establishes and displays a user’s personal availability and willingness to communicate through the use of common states such as Available or Busy. This rich presence information enables other users to immediately make effective communication choices. Conferencing Lync Server includes support for IM conferencing, audio conferencing, web conferencing, video conferencing, and application sharing, for both scheduled and impromptu meetings. All these meeting types are supported with a single client. Lync Server also supports dial-in conferencing so that users of public switched telephone network (PSTN) phones can participate in the audio portion of conferences. Conferences can seamlessly change and grow in real time. For example, a single conference can start as just instant messages between a few users, and escalate to an audio conference with desktop sharing and a larger audience instantly, easily, and without interrupting the conversation flow. Enterprise Voice Enterprise Voice is the Voice over Internet Protocol (VoIP) offering in Lync Server 2010. It delivers a voice option to enhance or replace traditional private branch exchange (PBX) systems. In addition to the complete telephony capabilities of an IP PBX, Enterprise Voice is integrated with rich presence, IM, collaboration, and meetings. Features such as call answer, hold, resume, transfer, forward and divert are supported directly, while personalized speed dialing keys are replaced by Contacts lists, and automatic intercom is replaced with IM. Enterprise Voice supports high availability through call admission control (CAC), branch office survivability, and extended options for data resiliency. Support for remote users You can provide full Lync Server functionality for users who are currently outside your organization’s firewalls by deploying servers called Edge Servers to provide a connection for these remote users. These remote users can connect to conferences by using a personal computer with Lync 2010 installed, the phone, or a web interface. Deploying Edge Servers also enables you to federate with partner or vendor organizations. A federated relationship enables your users to put federated users on their Contacts lists, exchange presence information and instant messages with these users, and invite them to audio calls, video calls, and conferences. Integration with other products Lync Server integrates with several other products to provide additional benefits to your users and administrators. Meeting tools are integrated into Outlook 2010 to enable organizers to schedule a meeting or start an impromptu conference with a single click and make it just as easy for attendees to join. Presence information is integrated into Outlook 2010 and SharePoint 2010. Exchange Unified Messaging (UM) provides several integration features. Users can see if they have new voice mail within Lync 2010. They can click a play button in the Outlook message to hear the audio voice mail, or view a transcription of the voice mail in the notification message. Simple deployment To help you plan and deploy your servers and clients, Lync Server provides the Microsoft Lync Server 2010, Planning Tool and the Topology Builder. Lync Server 2010, Planning Tool is a wizard that interactively asks you a series of questions about your organization, the Lync Server features you want to enable, and your capacity planning needs. Then, it creates a recommended deployment topology based on your answers, and produces several forms of output to aid your planning and installation. Topology Builder is an installation component of Lync Server 2010. You use Topology Builder to create, adjust and publish your planned topology. It also validates your topology before you begin server installations. When you install Lync Server on individual servers, the installation program deploys the server as directed in the topology. Simple management After you deploy Lync Server, it offers the following powerful and streamlined management tools: Active Directory for its user information, which eliminates the need for separate user and policy databases. Microsoft Lync Server 2010 Control Panel, a new web-based graphical user interface for administrators. With this web-based UI, Lync Server administrators can manage their systems from anywhere on the corporate network, without needing specialized management software installed on their computers. Lync Server Management Shell command-line management tool, which is based on the Windows PowerShell command-line interface. It provides a rich command set for administration of all aspects of the product, and enables Lync Server administrators to automate repetitive tasks using a familiar tool. While the IM and presence features are automatically installed in every Lync Server deployment, you can choose whether to deploy conferencing, Enterprise Voice, and remote user access, to tailor your deployment to your organization’s needs.

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  • Introducción a ENUM (E.164 Number Mapping)

    - by raul.goycoolea
    E.164 Number Mapping (ENUM o Enum) se diseñó para resolver la cuestión de como se pueden encontrar servicios de internet mediante un número telefónico, es decir cómo se pueden usar los los teléfonos, que solamente tienen 12 teclas, para acceder a servicios de Internet. La parte más básica de ENUM es por tanto la convergencia de las redes del STDP y la IP; ENUM hace que pueda haber una correspondencia entre un número telefónico y un identificador de Internet. En síntesis, Enum es un conjunto de protocolos para convertir números E.164 en URIs, y viceversa, de modo que el sistema de numeración E.164 tenga una función de correspondencia con las direcciones URI en Internet. Esta función es necesaria porque un número telefónico no tiene sentido en el mundo IP, ni una dirección IP tiene sentido en las redes telefónicas. Así, mediante esta técnica, las comunicaciones cuyo destino se marque con un número E.164, puedan terminar en el identificador correcto (número E.164 si termina en el STDP, o URI si termina en redes IP). La solución técnica de mirar en una base de datos cual es el identificador de destino tiene consecuencias muy interesantes, como que la llamada se pueda terminar donde desee el abonado llamado. Esta es una de las características que ofrece ENUM : el destino concreto, el terminal o terminales de terminación, no lo decide quien inicia la llamada o envía el mensaje sino la persona que es llamada o recibe el mensaje, que ha escrito sus preferencias en una base de datos. En otras palabras, el destinatario de la llamada decide cómo quiere ser contactado, tanto si lo que se le comunica es un email, o un sms, o telefax, o una llamada de voz. Cuando alguien quiera llamarle a usted, lo que tiene que hacer el llamante es seleccionar su nombre (el del llamado) en la libreta de direcciones del terminal o marcar su número ENUM. Una aplicación informática obtendrá de una base de datos los datos de contacto y disponibilidad que usted decidió. Y el mensaje le será remitido tal como usted especificó en dicha base de datos. Esto es algo nuevo que permite que usted, como persona llamada, defina sus preferencias de terminación para cualquier tipo de contenido. Por ejemplo, usted puede querer que todos los emails le sean enviados como sms o que los mensajes de voz se le remitan como emails; las comunicaciones ya no dependen de donde esté usted o deque tipo de terminal utiliza (teléfono, pda, internet). Además, con ENUM usted puede gestionar la portabilidad de sus números fijos y móviles. ENUM emplea una técnica de búsqueda indirecta en una base de datos que tiene los registros NAPTR ("Naming Authority Pointer Resource Records" tal como lo define el RFC 2915), y que utiliza el número telefónico Enum como clave de búsqueda, para obtener qué URIs corresponden a cada número telefónico. La base de datos que almacena estos registros es del tipo DNS.Si bien en uno de sus diversos usos sirve para facilitar las llamadas de usuarios de VoIP entre redes tradicionales del STDP y redes IP, debe tenerse en cuenta que ENUM no es una función de VoIP sino que es un mecanismo de conversión entre números/identificadores. Por tanto no debe ser confundido con el uso normal de enrutar las llamadas de VoIP mediante los protocolos SIP y H.323. ENUM puede ser muy útil para aquellas organizaciones que quieran tener normalizada la manera en que las aplicaciones acceden a los datos de comunicación de cada usuario. FundamentosPara que la convergencia entre el Sistema Telefónico Disponible al Público (STDP) y la Telefonía por Internet o Voz sobre IP (VoIP) y que el desarrollo de nuevos servicios multimedia tengan menos obstáculos, es fundamental que los usuarios puedan realizar sus llamadas tal como están acostumbrados a hacerlo, marcando números. Para eso, es preciso que haya un sistema universal de correspondencia de número a direcciones IP (y viceversa) y que las diferentes redes se puedan interconectar. Hay varias fórmulas que permiten que un número telefónico sirva para establecer comunicación con múltiples servicios. Una de estas fórmulas es el Electronic Number Mapping System ENUM, normalizado por el grupo de tareas especiales de ingeniería en Internet (IETF, Internet engineering task force), del que trata este artículo, que emplea la numeración E.164, los protocolos y la infraestructura telefónica para acceder indirectamente a diferentes servicios. Por tanto, se accede a un servicio mediante un identificador numérico universal: un número telefónico tradicional. ENUM permite comunicar las direcciones del mundo IP con las del mundo telefónico, y viceversa, sin problemas. Antes de entrar en mayores profundidades, conviene dar una breve pincelada para aclarar cómo se organiza la correspondencia entre números o URI. Para ello imaginemos una llamada que se inicia desde el servicio telefónico tradicional con destino a un número Enum. En ENUM Público, el abonado o usuario Enum a quien va destinada lallamada, habrá decidido incluir en la base de datos Enum uno o varios URI o números E.164, que forman una lista con sus preferencias para terminar la llamada. Y el sistema como se explica más adelante, elegirá cual es el número o URI adecuado para dicha terminación. Por tanto como resultado de la consulta a la base dedatos Enum siempre se da una relación unívoca entre el número Enum marcado y el de terminación, conforme a los deseos de la persona llamada.Variedades de ENUMUna posible fuente de confusión cuando se trata sobre ENUM es la variedad de soluciones o sistemas que emplean este calificativo. Lo habitual es que cuando se haga una referencia a ENUM se trate de uno de los siguientes casos: ENUM Público: Es la visión original de ENUM, como base de datos pública, parecida a un directorio, donde el abonado "opta" a ser incluido en la base de datos, que está gestionada en el dominio e164.arpa, delegando a cada país la gestión de la base de datos y la numeración. También se conoce como ENUM de usuario. Carrier ENUM, o ENUM Infraestructura, o de Operador: Cuando grupos de operadores proveedores de servicios de comunicaciones electrónicas acuerdan compartir la información de los abonados por medio de ENUM mediante acuerdos privados. En este caso son los operadores quienes controlan la información del abonado en vez de hacerlo (optar) los propios abonados. Carrier ENUM o ENUM de Operador también se conoce como Infrastructure ENUM o ENUM Infraestructura, y está siendo normalizado por IETF para la interconexión de VoIP (mediante acuerdos de peering). Como se explicará en la correspondiente sección, también se puede utilizar para la portabilidad o conservación de número. ENUM Privado: Un operador de telefonía o de VoIP, o un ISP, o un gran usuario, puede utilizar las técnicas de ENUM en sus redes y en las de sus clientes sin emplear DNS públicos, con DNS privados o internos. Resulta fácil imaginar como puede utilizarse esta técnica para que compañías multinacionales, o bancos, o agencias de viajes, tengan planes de numeración muy coherentes y eficaces. Cómo funciona ENUMPara conocer cómo funciona Enum, le remitimos a la página correspondiente a ENUM Público, puesto que esa variedad de Enum es la típica, la que dió lugar a todos los procedimientos y normas de IETF .Más detalles sobre: @page { margin: 0.79in } P { margin-bottom: 0.08in } H4 { margin-bottom: 0.08in } H4.ctl { font-family: "Lohit Hindi" } A:link { so-language: zxx } -- ENUM Público. En esta página se explica con cierto detalle como funciona Enum Carrier ENUM o ENUM de Operador ENUM Privado Normas técnicas: RFC 2915: NAPTR RR. The Naming Authority Pointer (NAPTR) DNS Resource Record RFC 3761: ENUM Protocol. The E.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM). (obsoletes RFC 2916). RFC 3762: Usage of H323 addresses in ENUM Protocol RFC 3764: Usage of SIP addresses in ENUM Protocol RFC 3824: Using E.164 numbers with SIP RFC 4769: IANA Registration for an Enumservice Containing Public Switched Telephone Network (PSTN) Signaling Information RFC 3026: Berlin Liaison Statement RFC 3953: Telephone Number Mapping (ENUM) Service Registration for Presence Services RFC 2870: Root Name Server Operational Requirements RFC 3482: Number Portability in the Global Switched Telephone Network (GSTN): An Overview RFC 2168: Resolution of Uniform Resource Identifiers using the Domain Name System Organizaciones relacionadas con ENUM RIPE - Adimistrador del nivel 0 de ENUM e164.arpa. ITU-T TSB - Unión Internacional de Telecomunicaciones ETSI - European Telecommunications Standards Institute VisionNG - Administrador del rango ENUM 878-10 IETF ENUM Chapter

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