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  • How do I setup a sip account with pennytel in empathy or ekiga in ubuntu 10.04?

    - by Glen
    Hi, I am trying to get my PennyTel VOIP account working in ubuntu 10.04. I know ubuntu use to use ekiga for sip calls. I also know that ubuntu now uses empathy for sip calls. However I can't get either program to work with my VOIP provider. In empathy I have no idea what details to enter. How does it know I want to use pennytel? What is my user name? I tired [email protected] but it did not work. In Ekiga I can't figure out how to use a sip provider. It looks like I can only use the Ekiga sip provider, which I don't want to use because I already have a PennyTel account. Any help would be appreciated. Thanks, Glen.

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  • Asterisk Register username with special character like "@"

    - by Najibul Huq
    I am using a SIP provider that has provided me with a username like: [email protected] (Note this is only the username part) And has a numerical password. My Register string looks something like this: [email protected]:[email protected] But this is not working, as asterisk is only sending the first part +112223344 before the first @. My provider is adamant about having the full form of it. This is the first time I am facing this issue that is quite unusual for me. Please help.

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  • Asterisk, IAXModem & Hylafax how-tos?

    - by Brian Postow
    I'm trying to set up Asterisk and IAXModem to send faxes via T38 (Yes, I know I'm swatting a fly with a Buick...) However, since I'm trying to do something so small with a product so large, I'm having trouble finding samples or how-tos that show me how to set this up. I've got all three installed, and I THINK I have my IAXModem config correct. I'm pretty sure that I have Hylafax correct (I've used it with T38Modem) so, I need to know which of the Asterisk samples I need to use, and how to use them. I think I want to use some combination of iax.conf, iaxprov.conf, sip.conf and sip_notify.conf. But I'm not sure where to put them, or what to change... I'm sure that the answer is RTFM, but I'm not sure WHICH M, or where in it to R... thanks. EDIT On a mailing list, someone told me that this actually WON'T WORK because IAX doesn't do T38. So, is there some other way to get Asterisk to work with Hylafax and send T38? I know that Asterisk does T38, the question is how to get the data from Hylafax and back...

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  • Setup wiped Polycom phone without SIP server

    - by Justin
    I'm troubleshooting a Polycom SoundPoint IP 550. I have wiped the hard disk of the phone (via a menu option) and now it's stuck in a reboot cycle. Apparently the only way to setup the firmware of the phone is to use a boot server. Does anyone know another way to setup the phone/firmware?

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  • sip.conf configuration file - add new line to each record

    - by Flukey
    I have a sip configuration file which looks like this: [1664] username=1664 mailbox=1664@8360 host=192.168.254.3 type=friend subscribemwi=no [1679] username=1679 mailbox=1679@8360 host=192.168.254.3 type=friend subscribemwi=no [1700] username=1700 mailbox=1700@8360 host=192.168.254.3 type=friend subscribemwi=no [1701] username=1701 mailbox=1701@8360 host=192.168.254.3 type=friend subscribemwi=no For each record I need to add another line (vmxten for each record) for example the above becomes: [1664] username=1664 mailbox=1664@8360 host=192.168.254.3 type=friend subscribemwi=no vmexten=1664 [1679] username=1679 mailbox=1679@8360 host=192.168.254.3 type=friend subscribemwi=no vmexten=1679 [1700] username=1700 mailbox=1700@8360 host=192.168.254.3 type=friend subscribemwi=no vmexten=1700 [1701] username=1701 mailbox=1701@8360 host=192.168.254.3 type=friend subscribemwi=no vmexten=1701 What would you say would be the quickest way to do this? there are hundreds of records in the file, therefore modifying all of the records by hand would take a long time. Would you use Regex? Would you use sed? I'm interested to know how you would approach the problem. Thanks

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  • PJSIP Custom Registration Header (iOS)

    - by Daniel Redington
    I am attempting to setup SIP communication with an internal server (using the PJSIP library), however, this server requires a custom header field with a specified header value for the REGISTRATION call. For example's sake we'll call this required header "MyHeader". From what I've found, the pjsua_acc_add() function will add an account and register it to the server using a config struct. The parameter "reg_hdr_list" of the config struct has the description "The optional custom SIP headers to be put in the registration request." Which sounds like exactly what I need, however doesn't seem to have any effect on the call itself. Here's what I have so far: pjsua_acc_config cfg; pjsua_acc_config_default(&cfg); //...Some other config stuff related to the server... pjsip_hdr test; test.name = pj_str("MyHeader"); test.sname = pj_str("MyHdr"); test.type = PJSIP_H_OTHER; test.prev = cfg.reg_hdr_list.prev; test.next = cfg.reg_hdr_list.next; cfg.reg_hdr_list = test; pj_status_t status; status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id); From the server side, there are no extra header fields or anything. And the struct that is used to define the header (pjsua_hdr) has no "value" or equivalent field, so even if it did create the header, how does it set the value? Here's the refrence for the header list definition: Link And the reference for the header struct: Link Any help would be greatly appreciated!

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  • how to start with a voip/public announcement project?

    - by Metiu
    The main requirements are: open source solution on Linux support P2P VoIP calls support presence support multicast VoIP announcements (and maybe some way of setting up such a "conference") preferably serverless (maybe the network can get split and I'd need to keep the functionality for all clients that still see each other) I tried looking at telepathy, in particular telepathy-salut, but it seems to be quite a new technology, so it lacks clear/good documentation and/or working examples. I'm also evaluating SIP (e.g. SofiaSIP), but it's working only if connected to a server.

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  • How to add g729 codec in Android application?

    - by juned
    i am developing a SIP application for making and receiving a call and i want to add the G729 codec in my application. currently i am doing analysis on open source project SipDroid. if i want to make that application to support G729 codec how to do that? there is a different codecs configuration file in org.sipdroid.codecs package.how do create the this kind of .java file for G729 codec? Any suggestion and response will be appreciated.

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  • IP telephony open source systems

    - by danke
    I'm trying to pick an IP telephony technology to learn. I heard of Asterisk, trixbox, freePBX, and my head was already spinning being not sure what to learn. Then I came across this article listing some more like Kamailio, Yate, CallWeaver, FreeSWITCH, SipXecs and now my head REALLY is spinning http://www.cio.com.au/article/323016/five_open_source_ip_telephony_projects_watch . Can someone give me a run down of how all these technologies tie together? What is the trend now, because I'd like to start learning. Note: Anyone please re-tag this question if you know better, because I'm new to this field and not sure about the best tags.

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  • Understanding Asterisk Server features

    - by Arham Ali Qureshi
    I need to ask few question about Asterisk 1) Does ACL mean by Access Control list here ?If yes than how could i use it? >ip show user 6001 * Name : 6001 Secret : <Set> MD5Secret : <Not set> Context : DLPN_Admin Language : AMA flags : Unknown Transfer mode: open MaxCallBR : 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 2147483647 Callgroup : 1 Pickupgroup : 1 Callerid : "test" <6001> ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No 2) What is mean by "Require Call Token" in Asterisk Digium GIU on Create new User Panel 3) Is There any command from where i can get users VOICE MAIL password ? 4) What AMI or CLI command set call recording on or off for user ? and if i want that file to be stored on client computer not on server memory what could i do ?

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  • Make and receive calls from and to PC to mobile and vice versa

    - by Hunt
    I want to route normal phone calls (i.e. calls made from landline or mobile) to VoIP and vice versa. Fr example, if I dial a number from a PC I will be able to call the other person, and the other person is able to see my number on their screen. Similarly, if a person calls me, I can pick up a call on my PC and can see their number on my screen. I don't have any idea how to implement this – how would I go about doing that?

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  • Asterisk Connection not working

    - by Tamas Ionut
    I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something like 192.168.x.y?? ). However, when I navigated to that url from outside of VirtualBox, that url pointed to nothing. Also I am logged in as root@localhost. Should I be logged in as root@server? I have also validated the installation as described here and everything went well. I am a complete beginner at Asterisk.

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  • Lync Server 2010 with Hosted VoIP PBX

    - by kmehta
    We just deployed Lync Server 2010 in our organization and it is working great so far. The next step for us is to enable enterprise voice so that we can replace our telephones with service that is handled 100% by Lync. This is where I am at a loss. I have a fully deployed Standard Edition Lync server and a hosted VoIP PBX provider with VoIP handsets. I would like to get rid of the handsets and have my company's phone service be handled by Lync client (e.g. someone calls my work number, and Lync rings instead of my old handset that is set up with the PBX) I am new to deploying these types of features. Any help is appreciated. Thanks.

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  • Asterisk dialplan context and label clarifications

    - by liv2hak
    I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the web.Can someone confirm if the IVR below is correct? Correct in the sense that if the below is used will it work.I know the IVR does not do much yet.But I am just trying to clarify my understanding. [incoming] exten => 123,1,Answer() same => n(menuprompt),Background(main-menu) exten => 1,1,Playback(digits/1) same => n,Goto(incoming,menuprompt,123) exten => 2,1,Playback(digits/2) same => n,Goto(incoming,menuprompt,123) exten => 9,1,Hangup() [main-menu] exten => n(menuprompt),Background(main-menu) exten => 3,1,Playback(digits/3) same => n,Goto(main-menu,menuprompt,n) exten => 4,1,Playback(digits/4) same => n,Goto(main-menu,menuprompt,n) exten => 9,1,Hangup()

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  • Customizing Sipdroid

    - by Seshu Vinay
    I have a Voip based app. So i thought of customizing SipDroid open source project. As the starting phase i have changed all the package names, Class names etc. It perfectly works on my Samsung Galaxy Y. But i tried with many other mobiles, Voice is not audible. Call is being initialized but could not hear voice. In Samsung galaxy young, voice is clearly audible for both incoming and outgoing calls. But in all the other phones i have tested i can hear only beep sounds. What could be the problem? I tried calling the other mobile(that has Sip Droid) with my Samsung Galaxy Young(customized app) I could hear voice in my app but could not hear in Sipdroid. But when i tried calling Sipdroid to Sipdroid, it works fine in all the mobiles.

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  • How to send REGISTER request periodically from my SIP client to Asterisk server using Asterisk Manag

    - by Prashant
    Hi, I am using Asterisk 1.4 server and I have created a desktop client using the Asterisk.NET Library. I am able to log into the AMI (as a manager) using Asterisk.Net, but I cannot find a way to send the REGISTER command using the AMI, to share my client's location information with the server. I want to know an AMI or a CLI command that can send a REGISTER request to the Asterisk Server. Thanks

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  • How to implement VOIP + SIP in iPhone?

    - by hib
    Hello all, I want to develop a VOIP application for iPhone . But I don't know the basics of VOIP concepts and also If there are any sources or library available that can I use in my application . So If anyone can provide me VOIP learning resources or library or anything that is useful in terms of VOIP and iPhone it will be precious to me . Thanks ,

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  • NAPTR support for SLES

    - by egiakoum1984
    Do you know if there is a library in SLES9 or SLES10 which support NAPTR queries for ENUM functionality? The libadns doesn't support NAPTR. There are other libraries which support NAPTR queries but they aren't included in SLES.

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  • Building a titanium module with xcode that uses other libraries

    - by kudorgyozo
    I have a Titanium module and i want to use it for voice over ip using pjsip. I have changed the project settings the following way: added to the other linker flags the libraries from pjsip added to the header search paths the headers from pjsip added to the library search paths the libraries from pjsip If i do these things for a normap iPhone app it works i can make calls i have tested it and made a wrapper class that has methods like makeCall, hangup. etc. But i want to use this class together with the libraries from pjsip in a Titanium module. It gives me errors like: implicit declaration of function 'pjsua_perror' implicit declaration of function 'pjsua_destroy' 'pjsua_config' undeclared (first use in this function) These are all part of pjsip (pjsua_perror and pjsua_destroy are functions and pjsua_config is a struct) Does it work this way? Can i include other libraries in a library? What is the difference between making an app that uses libraries and making a library that uses libraries?

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  • Using UCMA to connect to 3CX?

    - by Rodney Burton
    Has anyone used Microsoft's UCMA 2.0 SDK to connect to 3CX's free IP PBX to add voice capabilities to their application? If so, does it work? What I am trying to accomplish is having a windows form app running on 2 or more computers, and each person can connect to another person and carry on a voice conversation using their headset connected to their computer. App is in C# w/ .NET3.5 SP1.

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