AastraJobs

We are trying to build a new Asterisk server using PJSIP instead of the older SIP module and need some help to make it work. The setup is as follows: Asterisk certified/16.8-cert3 running on CentOS 8.2 with a public IP address. Phones in the office need to connect to the Asterisk server, they are on a private LAN (192.168.1.x/24) and access the Internet via a NAT router. Two types of phones: Aastra 6757i and Grandstream N300. The server also needs to send and accept calls over a SIP trunk to another Asterisk server also on a public IP (PSTN Provider). You will need to be an expert with Asterisk and PJSIP and work with us to get this working and explain what you did - you can have SSH access to the Asterisk server if required. For the right person this should be a fairly quick ...

...contractor to update configuration on our VOIP environment (a dozen or so phones). Currently using PiaF / FreePBX but happy to change to another distro if needed. Most phone are Mitel/Aastra 6739i and plus a couple cheaper aastra and a polycom conf phone. Key Issues / Targets: 1. Hot-desking - Configuration to allow for user speeddials to move with users as they log into their phones. 2. Call barging is not working properly at the moment. 3. Peering to another VOIP server 4. Bring our old server up to scratch. 5. A few more minor ones that I can't remember right now As the aastra phones are currently using the aastra XML login scripts experience on these handsets will be ideal. However will consider ditching the XML and use server-side endpoint ...

Hi, I require a VOIP service. Require a UK phone number, and also ability to call (package options?) I will be using a Aastra 6730i VOIP phone. Bid accordingly Thanks

I have a AAstra 6753i phone behind a NAT and a PBX hosted on a public IP. I am able to register another SIP device but not this phone. Need help to setup phone and extension properly so that port registration occurs on a dynamic port and not 5060.

I want to start VOIP Land phone business like ( ) which have DID Number , World Plus calling option for Unlimited Calling 75 Countries, Regular Home Phone Service for Unlimited US and Canada, The device is connected through the internet like Aastra, Grandstream,Linksys,ObiHai,Polycom or Yealink.,,,Free Softphone, Free Iphone APP,Having Caller ID and Excellent Calling feature whichever customer is attracted and I can Register 2 Cellphone number from another operator with customer can use my unlimited Minute for international call as my land phone as a access number.

We have subscribed to a Hosted FreePBX solution and need to configure it to work with our SIP Trunks. We also need to make sure this system is Absolutely Secure - Asterisk Security Expertise is required. The system has 3-4 sip trunks, 6 extensions (Aastra 480i based phone systems), 4 softphone extensions Need to setup one ring group and day/night/holiday auto attendant and call routing On-Hold music, Auto Attendant The built-In freepbx call administration - isymphony als needs to be configured. The system is installed - it just needs to be fine-tuned and configured to work with our Trunk Provider.

Im looking for somehow who could setup / configure an asterisk with Aptus FonB Integration for me. The Asterisk should be configured with one sipgate account and 1 phone. (Most propably an Aastra SIP 6739i) The FonBn should synronize 1 Google Contact account with the phonebook of the phone. A CTI module to dial from outlook should be installed as well. Which Asterisk Distribution is used is completly open as soon as it works. I can install the distribution, so its mainly about the configuration of the system.

=== Einsatzbereich === * Support bei spannenden VoIP-Kundenprojekten * Unterstützung bei der Weiterentwicklung unseres modernen VoIP-Telefonanlagen-Produkts === Must have === * Profi in der Systemadminist...Support bei spannenden VoIP-Kundenprojekten * Unterstützung bei der Weiterentwicklung unseres modernen VoIP-Telefonanlagen-Produkts === Must have === * Profi in der Systemadministration von Asterisk und Linux * Umfangreiche Erfahrungen in der VoIP-Optimierung von Netzwerken mit verschiedensten Firewalls (QoS, NAT) * Umfangreiche Erfahrung in der Provisionierung von VoIP-Telefonen (Snom, Aastra) * Deutschsprachig === Nice-To-Have === * Erfahrung mit den Firewall-Produkten pfSense und Fortigate * Erfahrung in der System-Administration von ISDN-S...

I need you to create the ssl certificates and setup freepbx and aastra and yealink ip phones so that they will communicate using TLS for sip and SRTP for rtp packages. Only bid if you have experience with an identical scenario. MAX budget is $40! Please do not place higher bids if you want to be considered. Thanks

Hello, We have a FreePBX Asterisk server behind a pfSense hardware router on a 10MB symmetrical internet connection. Everything seem to be connected and setup properly, our close to 15 users all use hardwired Aastra handsets. I would like an experienced hand with narrowing down call quality issues that include bad echo, cross talk, delay when answering or call (outbound calls don't always hear it ring, and there's sometimes a delay with inbound and outbound calls when they are first answered). Our VOIP provider insists it's not an issue with their services. All logins will be provided, project must be completed within 24 hours.

I have a elastix install with 3 aastra 6757i phones. One phone has the expansion panel. I have the xml scripts installed and calls can be moved round to all phones and put on hold/xfer/park. I am looking for consulting / configuring call flow and fixing a few of the minor bugs I have found in my testing. Call Flow: What is the standard way you put a person on hold? Queue / Parking Don't care just want to be able to move calls round easy Bug Examples. Music on hold streaming. Presence feature to show on expansion panel DND to show on Expansion panel If you have experiences with the phones or elastix you might have already have some of these items fixed. We would like to add this system in to our website and bring in other on board other features ov...

Need a experienced XML programmer with VoIP experience. Need xml file written for Aastra phone 6757. Need a pop-up message screen for Barge-In call. Currently when barged in to a phone call, Aastra phone screen lights up, but no message. There is a Barge in sound, however need a message on a screen also.

Simple and short request. I need assistance in building a Configuration file for the AASTRA 57i phones. The phone downloads via TFTP/FTP the configuration file from the server. The config file contains the extensions, and programmable features. Without the config file, I have to manually program each file. ## Deliverables requirements notice: As originally posted, this project does not have complete details. Should a dispute arise and this project go into arbitration "as is", the contract's vagueness might cause it to be interpreted against you, even though you were acting in good-faith. So for your protection, if you are interested in this project, please work-out and document the requirements onsite.

New business inherited Aastra phone system with mostly 6757i and older phones. A dry DSL line enters the building and is hooked up to a small computer with FreePBX distro installed and running. Need help setting up the SIP trunks and Aastra extension so that 4 phones in the building work, need help ASAP, feel we're close but need help to finish. Knowledgeable network individuals only please with Asterisk and/or FreePBX experience. Payment on success.

We are considering establishing our own auto provisioning server for the voip phones that we sell We sell: Aastra Cisco Gigaset Grandstream Polycom SNOM Yealink Most if not all have provisioning redirection services that would allow us to redirect phones with specified MAC addresses to look to our auto provisioning server for config We are looking at Open Source at the moment. Maybe someone with existing experience and previous project work could provide us with a cost effective solution

...parts of the VoIP system to be actively maintained by the developers (e.g. Asterisk or FreeSWITCH, /not/ trixbox). We have a dedicated Internet connection strictly for VoIP, no landlines, 11 Aastra phones and less than 10 DIDs. You will be provided remote access to a dedicated freshly installed Linux server. Your job is to configure the VoIP system and either provide simple instructions to configure for Aastra's (providing support if needed on configuration) or set up some way for the Aastra's to automatically configure themselves (e.g. a TFTP server) Required features: Support up to 15 Aastra 9133 SIP phones, each with its own extension Support our Linksys-based analog-to-VoIP adapter for the office speakerphone Interface to our existing VoIP provid...

We have a production server with 500+ accounts using linksys ata's and a few hosted PBX clients with cisco and aastra phones and a couple iax trunks. We need an asterisk expert to look over our logs asterisk configuration/dialplans etc. to review/find and resolve some issues that we cannot find. The Details Running Asterisk 1.4.23.1 Gentoo Asterisk sometimes crashes for no apparent reason. There is nothing in the logs to suggest that anything out of the ordinary was happening at all. - Asterisk sometimes crashes when people are checking voicemail and deleting several messages in quick succession. This is likely a buffer overflow in the Voicemailmain() command. - Calls drop at random for no reason that we can discern. In our office, we have ...

We need a unified portal for configuring and provisioning our Asterisk based VOIP system. This portal will include PBX configuration (managing DIDs, Trunks, Routes, etc.), as well as VOIP phone provisioning/configuration (Polycom and Aastra primarily). It may integrate or utilize existing tools like FreePBX (which is currently being used to administer the system). VOIP/Asterisk knowledge and experience will be essential. The portal should also be extensible and scalable. We will eventually need to add other services to it, such as Hosted Exchange Mail, spam filtering, etc via the respective provider's API. It will also be managing several IP-PBX phone systems, so the portal should be able to provision services to any of the several servers. The portal should be funct...

We need a unified portal for configuring and provisioning our Asterisk based VOIP system. This portal will include PBX configuration (managing DIDs, Trunks, Routes, etc.), as well as VOIP phone provisioning/configuration (Polycom and Aastra primarily). It may integrate or utilize existing tools like FreePBX (which is currently being used to administer the system). VOIP/Asterisk knowledge and experience will be essential. The portal should also be extensible and scalable. We will eventually need to add other services to it, such as Hosted Exchange Mail, spam filtering, etc via the respective provider's API. It will also be managing several IP-PBX phone systems, so the portal should be able to provision services to any of the several servers (Including interconnectivity betw...

Need remote assistance in the migration of taking a Private School from an older analog phone system over to PBX In A Flash (PIAF) VOIP System with Aastra Phones. I'm familiar with phone systems, networking and VOIP but not with PIAF. I will require documented steps of the options changed after the default installation to get the trunks and extensions working. Will also need support for a few days after the system is working to ensure the internal user's needs are met. The school's current phone system is very simple so I don't foresee very many customized features needed. ## Deliverables requirements notice: As originally posted, this bid request does not have complete details. Should a dispute arise and this project go into arbitration "as is", th...

...a few steps, connect the sip device to the internet, enter the MAC address of the sip device (this is printed physically on the label that is usually located on the bottom of the device) and select the device type in the auto provisioning server. The AP server must be able to be in compliance with the following SIP device: • Polycom IP Phones (Text files) • Snom IP Phones (XML files) • Aastra IP Phones (Text files) • Broadcom Based IP Phones (Text files) • Linksys and Grandstream IP Phones & ATA The scope of work is to plan, deploy, test, and customize an Auto Provisioning platform to be able to be used by the above SIP devices. In addition, documentation of the final product, training and 30 days maintenance with an option of monthly ma...

We are looking for a VoIP expert to help build a growing VoIP network across North America. Some duties will be in designing and growing the network as demands increase and troubleshooting end to end issues as they arise. Here are the skills needed... 1. Understanding of SIP protocol 2. Understandi...servers / applications 4. Experience in configuration and troubleshooting SIP devices such as hardware phones, soft phones, and video devices. 5. Advanced trouble shooting skill in VoIP, networking, and related hardware Recommended Skills/Experience... Operating systems - Linux, Windows Switching - Broadsoft, Cisco, Opensips, Asterisk, Freeswitch, Mitel, Avaya, etc Devices - Polycom, Aastra, AudioCodes, Cisco, Snom, etc Networking - Firewall/Routers, Managed Switches,...

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We are looking for a full time VoIP (SIP) expert to help build a growing VoIP network across North America. Some duties will be in designing and growing the network as demands increase, and troubleshooting end to end issues as they arise. Skills Understanding of SIP protocol Understanding of IP ...switching servers / applications Experience in configuration and troubleshooting SIP devices such as hardware phones, soft phones, and video devices. Advanced trouble shooting skill in VoIP, networking, and related hardware Recommended Skills/Experience Operating systems - Linux, Windows Switching - Broadsoft, Cisco, Opensips, Asterisk, Freeswitch, Mitel, Avaya, etc Devices - Polycom, Aastra, AudioCodes, Cisco, Snom, etc Networking - Firewall/Routers, Managed Switches,...

Potrzebny spec od Trixboxa. Projekt sklada sie z 2 czesci : 1) ustawienia trixboxa , ustawienie polaczen na zewnatrz jak i przychodzacych, ustawienie kodekow, itp. jak i podanie konfiguracji do routera i do telefonow Aastra i Linkys aby dzialaly zarowno w konfiguracji LAN (w tej samej sieci co Trixbox) jak i z zewnatrz (przez NAT). 2) napisanie komponentu na Sugar CRM , ktory sluzylby jako predictive dialer uzywajac trixboxa. Opis funkcjonalnosci : - system wykonuje 3-5 proby polaczen do numerow z wybranej listy kontaktow z Sugar CRM wg wybranych parametrow bazy (np do kogo kontakt przypisany, itp, status jego itp) - w momencie kontaktu z zywa osoba, pozostale polaczenia sa przerywane i system automatycznie laczy z telemarketerem do ktorego kontakt jest przypisany (wg je...

...where you've looked, and try to include a similar picture instead or one of the logo of the company making it at least. The background of the images should be transparent, or white if transparent is not possible. ## Deliverables 3Com VCX Aastra Ascotel Intelligate A150 Aastra Ascotel Intelligate A300 Aastra Ascotel Intelligate 2025 Aastra Ascotel Intelligate 2045 Aastra OpenCom X320 Aastra/Ericsson Business Phone + M600 GW Aastra/Ericsson MD Evolution Aastra/Ericsson MX ONE Aastra OpenCom 130/150 Aastra OpenCom 510 Asterisk Auerswald COMpact 5020 VoIP Avaya CM/SES Avaya Aura SM Avaya Quick Edition Avaya IP Office Broadsoft Broadworks Cisco CCM Cisco CUCM Ci...

This project involves maximising the features of Kamailio. I have an integration between kamailio and asterisk currently and would like to get the most out of it. The main feature that needs to be working is presence. I would like all common ip phones such as polycom, linksys, snom, aastra and yealink to be able to use their BLF buttons via registering into Kamailio. If you know how to get this working and you are a real expert on Kamailio then I will pay extra for your knowledge. ## Deliverables Kamailio Presence Call Pickups in Kamailio Kamailio behind a NAT

Hi, snom phones, as cisco, aastra and so other, have his own XML mini-browser emulator where somebody can develop his own applications. Unfortunately, in order to develop/test the application, you need, of course snom phone. I would like to develop a windows based "snom emulator", only for the XML display issues, that can be used to emulate the screen/display of a snom phone (mainly 820 & 870 models, but models with fewer display lines, as 300 & 320, would be also a plus). I would need somebody to rewrite something similar to "the snom emulator display" that can be find here -> , I would prefer that the display is showed on a specific aplication (and not only parsed on the browsers as is on the link above)

Looking for asterisk/...with integrated prioritization for voip . ## Deliverables 1 customized boot image? 2. deployment to new server via usb stick or iso image , cd? 3 contain recovery partition where customer can restore original status and eventually restore previously backed up extensions dial up plan? 4 shall include all in one appliance with routing and qos software . VoIP prioritization? 5 animated aastra 57i logo? 6 include profile for lcd expansion module? 7 updated a2billing? 8 new elastix theme? 9 posibility to update installation media to latest centos elastix packages? 10 posibility to backup recordings to external media or add storage capacity via usb external media? 11 the customized image shall include vmware tools and able to work in vmware e...

we are using aastra 480i and 57i phones with trixbox 2.6 visual voicemail is not working

We want to automatically download our outlook contacts and make them appear in aastra 57i directory We are running trixbox ce 2.6

I have aastra 57i working fine with Asterisk server from "PBX in a Flash" with tftp provisioning. I tried to use the same with 53i but I can not get Intercom to work. I think we can do provisioning with HTTP instead of TFTP from asterik because it looks like needs to be different for each phone model. Thank you,

Hello All, We have a call center where users use Aastra 53i. Right now, we have Aastra 53i setup wit endpoint xml to be able to log in any phone. We also have static agents in the Queue. However, this is not desired anymore. We will setup dynamic agents for Queue and want payroll system. Eveytime an agent logs into a Queue, their extension # and time (EPOCH) will be saved in MySql database. Everytime agent logs out, the same thing happens. Finally, all this data is sorted and displayed on a password protected page where the Admin can browse and see how productive the users were. The following fields are important: a- Who worked for the day. b- How many minutes or hours they were logged in out of 8 hour shift. c- Most productive to least productive users are sorted out...

have a list of 2000 personal names and the company name. I need the business adress and business phone number inserted in an excel file. excel file should include *company name *person name *street *zip *City *Telephone *company website All Adresses reside in Germany Europe and can be found on german google www.google.de...*company website All Adresses reside in Germany Europe and can be found on german google or german phonebook Time to complete 5 days !!!!offers only with sample research!!!! Sample information for sample research: Olaf Fieber Kryolan GmbH Stephan Leder Kieback & Peter GmbH & Co. KG Guenter Haeublein readybank Andreas Szameit Aastra DeTeWe GmbH Michael Thoss DRK Kliniken Berlin

...Alarms, Framing, Signalling, etc. -Analog -SIP -Ulaw, Read a trace, understand a 503, 488, 404 etc. -IAX2 -MySQL -Basic queries, delete/add records -Flash Operator Panel -Customize & Knowing the following is a plus (if you don't, you'll need to learn them) -Edgewater Edgemarc Devices -Elastix (extensive) -ViciDial (extensive) -Aastra XML (basic) -Postfix (basic) -Jabber/OpenFire (basic) -SNMP (basic) ## Deliverables Looking for an experienced tech AND one that would be prepared to do plenty of independent learning (non-billable) This is often a high stress environment that requires the ability to work under pressure and outside of normal business hours. You'll need to ...

If you have done this before and have the files then even better! We are looking for a MAC provisioning System. has to be; -User friendly and easy to user -Keeps MAC addresses and provides some reports -Stores Many different kinds of SIP Firmware for the following IP Phones ALL CISCO 7900, LINKSYS, AASTRA, AVAYA (of course ATA's as well) Will require an intergeneration to CRM as well... Thanks for your bid!

Hi, Does any wants to write/sell me a Phone Provisioning Script? I have following requirements. 1. TFTP Server 2. Web GUI to manage phones auto provisioning. (internal use) 3. The phones that will be used with this TFTP are the following: * Aastra 57i CT * Cisco 7940, 7960, 7970 and 7971 * Polycom 330, 550, 560 and 650 This is phone models we are supporting. here is how GUI should be functional We login to portal * user name * password then we have an option to browser phones by MAC address or by Client Name (company name) only for internal use. Add a phone option: * Select a brand name of the phone * Select a model * Enter a MAC address of the phone. * Enter SIP user name * Enter SIP user password * Enter SIP U...

I'm not sure if you'd be able to help. I have installed a 13 extension PBXNSIP product with two domains. One domain has 3 extension and the other has 10. I have or will be using Aastra and Snom SIP phones. Need to finalize installation and test to see if working properly. Will give access to server via VNC. Although I have two domains, I need help troubleshoot only 1 with 10 extensions. Only one of the 10 extention is active. I will add extension as need arises. I'm not sure if it's nescessary to create 9 dummy extentions to facilitate performance testing. I will need to setup Auto Attendant and User Agent for the ONE domain. Will allow server access via VNC. Please include sufficient details in your response for evaluation Thanks...

We need to implement a hot-desking solution whereby a user logs onto a Windows (XP/2003/Vista) PC and the telephone connected to that workstation (Aastra 9133 or 480i) is automatically configured with the users' extension number etc. Back-end PBX is currently Trixbox 1.2. The solution must work with Trixbox versions 1.2 upwards, including v2.0. Windows code must work in a .Net Framework 3/WCF environment (we can test this) but should not rely on .Net 3 for functionality. We have a SQL Server database containing user accounts, extension numbers and (will add) a field to indicate whether that user is eligible for hot desking. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Delive...

...our entire presentation completed. A portion of the deliverables includes making the separate graphic available. The server in slides 3 and 4 should be rack mountable and slick looking. We have two types of phones in the slides. Examples of what our phones look like can be found at: <>. The analog phones should look like the Scitec, Cybiolink, or Aastra. The IP phone should look like the Uniden or Polycom phone. Use your imagination. The five slides should have a unified, professional look. When bidding, please include past PowerPoint slides that you designed. If you desire, feel free to submit a screen shot of a mock slide to give us an idea what you might be doing. If you want to just do one or two slides, please specify in your bid. As I