similar to: SIP Proxy

Displaying 20 results from an estimated 1100 matches similar to: "SIP Proxy"

2012 Jan 03
4
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>
2008 Jun 12
2
Reg. setting Domain name on Cento 5 pc
Hi all, I am running centos 5.1 and I wish to change the domain name and dnsdomainname of my PC. currently the settings are-- $ hostname sipx.com $ hostname --fqdn sipx.com $ domainname (none) $ dnsdomainname com I have searched in the net for tips but everywhere only the hostname change is provided. I need to change/set the domain name and the dnsdomain name on my pc to sipx.com and this
2009 Mar 16
3
Asterisk is not designed for University with large user base?
Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running
2003 Sep 02
2
STUN server from Vovida
Not sure if it's alright to talk about this here??? compiled the STUN server from Vovida on RedHat 7.3. Looks simple to configure. It isn't starting...it tries to for a long time and then just craps out. Here is my config:/etc/sysconfig/stund #!/bin/echo Not to execute. # Path to stund STUND=/usr/sbin/stund # Set the required args for STUND STUNDPRIMARYHOSTNAME=208.x.x.x # The hostname
2004 May 04
7
stun server
What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?
2003 Apr 30
1
Buzzword bingo: TLS and SRTP
One of my clients today asked me about TLS support for encryption of SIP payloads, and I didn't have an adequate answer as to why it wasn't supported or even discussed. Some archive searching finds scant mention of this in reference to Asterisk. Of course, encrypting the SIP payload is only 1/2 the problem; the payload itself is the next problem. I understand that IAX solves these
2008 Sep 22
2
Newbie: Get echo cancellation level
Hi: I'm using speex to perform echo cancellation in Windows. I'm aware of the problem about out of sync clocks in record and play sample rates in usual sound cards . In order to have an idea of how good is my echo cancelation working I would like to know if there is any #define thing i can pass to speex_echo_ctl to get the actual level of echo cancellation. If not, how can i extract that
2003 Jul 14
1
Fwd:[Vocal] Question about Cisco IP hard phones
Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to
2005 Oct 16
2
Looking for advanced consultant services
Hi, I have a meeting with an important customer in a couple of days and I am aware that most of their questions are going to be related about scability of Asterisk. We want to propose this customer to integrate Asterisk with SER, but I have a loot of complex doubts that I would like to known before this meeting. I would like to contact with a busines that has experience with large
2005 Jun 28
3
PESQ results for speex 1.0.3
Hello! Some time back, I added the Speex protocol to my version of VOCAL (www.vovida.org, VOIP tool). Recently, I also added PESQ (automated voice quality testing algorithm) to my tool and have been running some tests on a clean network. The source file is a woman reading some phrases meant to test various aspects of codecs... Speex has a respectable result of 3.67 Some other codecs I've
2003 Sep 04
1
Asterisk vs. Vocal (Vovida) vs. Bayonne
Folks, I love Asterisk, have been using it for a while now. I'd like to know if anyone has some good comparison points on Asterisk vs. Vocal (Vovida) vs. GNU Bayonne. I know only a little about the later two. Also, one drawback I've hard about Asterisk (not for me, but for general consumption/deployment) is easy of configuration -- people like GUIs. They want point-n-click. I'm a
2008 Feb 05
6
External MWI question for Asterisk
Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into
2005 May 31
2
Sipura 2000 behind NAT issue, Vonage is working
Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear
2004 Sep 14
1
Comparisons between * and sipXpbx (PingTel's open source product)
Has anyone compared * to sipXpbx? From a cursory look, this open source version of PingTel's PBX has many features that make it more suitable as a replacement for a traditional PBX, including the ability for users to tell if a phone/trunk is in use. What I am trying to figure out is what I'd give up using sipX instead of * (and vice versa). /carmi
2010 Feb 26
3
: PSTN calls
Hi All, I have installed astriesk 6 and am able to make calls using sip x-lite. Its working as I expected. Now I want to make call from sipx-lite to PSTN using asterisk. can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.). 2) what is that I need to do after buying the card to make it talk to the real world PSTN network?
2003 Oct 24
1
Asterisk ???
Asterisk will become a real ip tel softswitch or is going to other way ? like vovida .... regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031024/361291f9/attachment.htm
2005 Mar 11
2
Load Balancing b/w 2 asterisk servers using SIP load balancer
Hi, I'm trying to do load balancing between 2 asterisk servers using SIP load balancer, provided by http://www.vovida.org I used the following options on lbproxy, but I get the below message continuously. ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2 "No proxies are up - can not send message to anyone" Xlite is not able to register to the
2007 Apr 15
2
Custom CentOS5 DVD
Hello, Does anyone have an up-to-date page describing, step by step, how to make a customized CentOS5 DVD? I noticed that CentOS5 already comes with ~240MB of updates. So for starters, I'd like to create a new DVD with all the current updates. (And I have other custom scripts I need to install on top of that). I've googled around and tried various suggestions on the net:
2007 Aug 29
0
re:Cisco cfgfmt.exe tool
Hi I am trying to your wine with sipXecs. We are testing Cisco ATA-186. When we attempt to update the ATA software via sipX we send a text file to the sipX server but it needs to be convert from a text file into a binary via the cfgfmt.exe tool I installed sipX and wine on the same box and used wine config tool to point or grab the tool. Wine puts it or a copy of it in the system32 folder.
2003 Oct 15
1
SER vs STUND with Asterisk..
One for the gurus.. I have seen there has been a lot of discussion about using SER with Asterisk.. This to me seemed like an over kill becasue it would basically be doing most of what Asterisk is doing anyway unless you create some weird and wonderful config in SER.. Anyway, I decided to go and have a quick read through the SER docs and in the section about NAT they say that the best way to