search for: sers

...ke: auth_radius.so and others. Can somebody help me with this issue?? Thanks Daniel. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 3612 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050525/11ba21cf/attachment.bin

...ring calls through asterisk, it seems that, for some reason, things get worse when SER is an intermediary in the communication. Could anybody help me here, please? Victor. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050808/438cf815/attachment.htm

Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in

Hey gang, I'm trying to work out all possible scenarios using SER & Asterisk in our upcomming deployment. The example scenario is 50 different customers, all with different numbers of SIP UAs. All UAs would register with SER; This will help keep any inter-office conversations off our bandwidth since SER doesn't handle the RTP stream. Calls from PSTN to UA are easy to handle.

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

...cs. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/30/2004 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040812/64838868/attachment.htm

Dear Mr. Beginning to work on Linux. I am trying to install R into Ubuntu Gustsy. I installed version 2.5.1 and worked fine. But then I tried to upgrade by adding these lines to sources.list: deb http://cran.fiocruz.br/bin/linux/debian etch-cran/ deb http://cran.fiocruz.br/bin/linux/ubuntu gutsy/ I introduced the authorization keys (as recommended) and the files were downloaded. But then I

...ave some problems to hear the other side. Thanks for any advice --------------------------------- Do you Yahoo!? Yahoo! Small Business - Try our new resources site! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050323/20839c61/attachment.htm

Hi, I am puzzled with serialize(). It comes down generating identical hash codes for (apparently) identical objects using digest::digest(), which in turn relies on serialize(). Here is an example illustration the issue: ser <- function(object, ...) { list( names = names(object), namesRaw = charToRaw(names(object)), ser = serialize(names(object), connection=NULL, ascii=FALSE)

...from Asterisk? Thanks and Regards Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040531/86d2abbc/attachment.htm

...of references to SER. Currently, I have: 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I have yet to open my server to the Internet to be accessible to travelers without VPN first. I have done some testing with VOIP provider though my firewall to FWD and VOIPSTUNT. Where might SER help? Why are people using...

Hi all, Hope somebody can help-I really am stumped as to why this won't work. I realise that this isnt an Asterisk problem (Please dont bash me on the list) and I have emailed the SER list but I havent received a reply and maybe someone on this list can help...Once this problem is solved I am going to use Asterisk for voicemail etc with SER (I have it set up) I currently have SER set up and

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

Why is SER considered a better SIPserver than asterisk , why is it that SER can handle more clients than asterisk can. And if this is just cause of say poor SIP handling code in asterisk then is there anything being done to fix it. Just wanted to know why SER claims to be better than asterisk as a SIP server. ? -- regards Vikram (http://www.vicramresearch.com)

...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 Message ----- From: braincrew.com To: serusers@iptel.org ; asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 5:00 AM Subject: [Serusers] ser+asterisk - security Hi there, I'm using ser and =sterisk together. Asterisk for voice mail etc and ser for registration of the =sers usig database. I can restrict =orwarding...

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

...am with lots of questions and confusions in this particular process hope some body will solve my headache confusion ..Thanks in advance Kindly regards, Ravi. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060601/93f9a5d6/attachment.htm

...tried the same scenario with an Asterisk 1.0.9 (without virtual and without real-time) on a Fedora Core distribution, and it works. Any idea? Best regards, -Stefano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071119/775a0b59/attachment.htm

...---- (Asterisk servers) (10.254.254.x) Phone A can belong to Asterisk A, and B to Asterisk B. Hope this give you enough information. Regards, Ronald Voermans -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050819/53aadd31/attachment.htm

I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in