similar to: IPv6 support

In the setup I'm looking now there are IPv4-only and IPv6-only SIP-clients (or UA's) but the Proxies (or servers) are v4 and v6 dual-stacked (to make things a bit easier). I think there will be a lot of IPv6-only SIP-clients (e.g. mobile clients and all the existing IPv4-NAT'ed networks) and a lot of IPv4-only "old legacy" SIP-clients. We have to find ways to deal with such

Hi, I've been experimenting with an IPv4 and IPv6 VoIP setup using SER. I'm using Asterisk for voicemail, etc. but as this only works for IPv4, I had to do a number of tricks to get it going for IPv6 phones. I was wondering whether there is any interest or plans in the pipeline to have Asterisk IPv6-enabled. Any info, especially by the developers out there, would be welcome. Thanks, --

Hi! I'm using asterisk with a few kphone SIP-Clients. The registration process seems quite OK. But there are some problems: Calling other registered users is possible, but the rtp-stream is not reaching the right port, so you can hear nothing. In ethereal you can see, that the SIP/SDP fields addresses different ports at each client, so client A sends to port 32000 but client B listens on

Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a "front end" for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or

hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

Alguien conoce este un test no paramétrico que se llama de Brunner-Langer model para dependent data, también conocido como modelo longitudinal? Intuyo que es como un Friedman pero que admite algún efecto fijo. Graccie mile. -- Antonio Maurandi López Universidad de Murcia Edif. CAID. Campus de Espinardo. 30100 Murcia @. amaurandi en um.es T. 868 88 7315 F. 868 88 7302 www.um.es/sai

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

Dear All, I have the following setup. Quad T1's<->Asterisk (PBX)<->(LAN<->DMZ)<->SER<->(Firewall)<->(Internet) | Local US Help Desk (Snom 200') This setup works well. I can pass calls from over the internet to the Asterisk PBX via SER using X-Ten Lit. I have a couple of questions; 1. How do I

Hello, I try set Ua---SER----Asterisk (voicemail/ARA) | Ua ser stable asterisk cvs head I read http://mail.iptel.org/pipermail/serusers/2005-February/015997.html to forward unavailable or busy sip agents to asterisk voicemail in failure route. How may I configure extensions.conf and ser.cfg ? I have been trying without success! Regards Harry

Hi all, I was wondering if someone could explain the relationship of Asterisk, SER & Jabber (XMPP) to me. I understand that there are facilities within Asterisk to use jabber to notify of incoming calls via XMPP clients, however I'm trying to work out exactly where the SER server would sit in all of this and what it's actual role would be. This might seem like a silly

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we

I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently I"m using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call

On 09/03/2016 10:56 AM, Etienne Dechamps wrote: > C will still need keys in order to establish metaconnections with A and B (as > well as a few other things). However there is no need for C to own any > "Subnets" at all. If somebody breaks into C, he could get access to the vpn network, right? Because the keys are there, it will be possible to use them to get access. Even if

I have coded chan_sip.c so that you can have // sip.conf register => username:password@somedomain.com/redirectconfig [redirectconfig] redirect=yes redirecturi=sip:12345@domain1.com redirecturi=sip:34556@domain2.com redirecturi=sip:87877@domain3.com .... so when you receive a call it will redirect to the alternating uri's with a SIP 300 Message. It works with the following sequence,

What's wrong with ARA (asterisk realtime architecture) from voip-info: Asterisk, SER and MWI http://mail.iptel.org/pipermail/serusers/2004-December/013727.html Actually I wrote a patch for this and it supports ast_data too. What you do is tell asterisk that all of your phones IP addresses are your SER machine. Then when a message gets left Asterisk sends the NOTIFY to username at

Hi. We are trying to set up asterisk to service a wireless community in our town. We have about 30/40 wireless working nodes each one with a 10.34.x.x/24 subnet for users. Each one of these addresses can potentially have a 192.168.x.x/x subnet. On top, the wireless nodes, themselves, are linked in 172.16.x.x/x subnets. On top of the top, there is internet and cool things for people, like

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

On 09/02/2016 08:51 PM, Etienne Dechamps wrote: > What version of tinc are you using? tinc 1.1 already does what you want out of > the box: packets sent from node A to node B through node C will use a key that > A and B will negotiate between themselves. C doesn't have the key, and will > act as a blind relay. C will not be able to decipher the packets flowing > between A and B.