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Dear all, I'll implement a VoIP system using Asterisk + SIP with softphones; I need to connect LAN and VPN users (about 100-150). What version/installation of asterisk do you recommend tyo me ??? Does Asterisk@Home or Trixbox match to my scenario ???? By the way, I use Debian Etch as OS server. Really thanks. Alejandro --

Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP with Linux/Debian Etch??? I'd like to see if my intranet contacts are available, busy, disconnected.... Thanks a lot Alejandro

I have Asterisk 1.2.13 installed as a Debian package and I edit only sip.conf and extensions.conf in this way: sip.conf: [general] realm=work.com.ar ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes

Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online,

People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with "apt-get" the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? 2) Do I have to configure a dummy PSTN interface in my case ?? And if you have a debian-asterisk howto, I really thank you. Regards,

Dear list, I have an Asterisk 1.4 SIP server and several voip clients in my LAN using Twinkle 0.9-6 and X-Lite, talking among them. I see the following features and I have the corresponding questions: 1) X-Lite 3.0 have SPEEX codec support but only for 14Kbps to 23Kbps bitrates. Can I set a lower bitrate here, i.e. 8Kbps ??? 2) Twinkle has SPEEX support too, and I think it's more manageable

People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on "localhost:8080", but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ???

People, I've setup Asterisk in a basic mode with SIP protocol. In the future I wanna connect several offices each one with an own Asterisk server, using IAX because I read it has no firewalling problems using just one UDP port for control and data -aming other advantages- . SIP has NAT problems I know. Do you recommend the use of IAX instead of SIP for users and among several Asterisk's

Dear all, I have an Asterisk server with SIP and IAX softphones clients, and I need to encrypt the voip calls among them: *For SIP clients I use Twinkle which implements the ZRTP/SRTP encryption mechanism client-2-client; I read it's the better security mechanism nowadays created by Phill Zimmerman who created PGP. *For IAX clients I used Kiax but I don't know exactly if there is any

Hi people, I have Asterisk 1.2.13/DebianEtch as my VoIP server, using SIP. I need to use IM (instant messaging) among X-Lite clients, but when I send a message to any other client I get the error "Error: method not allowed". I read Asterisk does not support instant messaging, so.....What's the best way to have instant messaging with Asterisk ??? Thanks a lot. Alejandro

People, I have an Asterisk 1.2 server and a Jabber server in different hosts. I need to implement voip+presence+instant messaging knowing that Asterisk does not support presence+IM.....So is it possible to use a softphone client (Gaim, X-Lite, etc.) to give to my users voip+presence+IM connecting to the Asterisk and Jabber servers at the same time ??? Thanks a lot Alejandro

Dear all, I have Asterisk 1.2.13 running OK with Twinkle clients, they can talk very well using SIP. I have a Jabber server running OK and the clients use PSI client for chat succesfully. Now I'm using Wengophone 2.1.1 in order to unify voip+IM services. The users can logon OK in SIP and Jabber, they get the online status presence, but they CAN'T talk and chat among them.

Dear all, I have Asterisk 1.4.13 and I need to use encryption among Asterisk and my SIP users, and with the RTP data interchanged among users. I prefer the use of ZRTP/SRTP because we use Twinkle and X-Lite/Zfone as our voip clients and they support these encryption mechanism. My question is: do I have to enable any encryption support in Asterisk 1.4.13 ??? Or Asterisk has native encryption

Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in "allow=gsm" line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF

Dear all, I have to implement a linux/iptables firewall between my SIP clients and the Asterisk 1.4.13 SIP server. There is no NAT in my implementation, so in sip.conf I have "canreinvite=no". I have iptables 1.3.6 version. Does iptables need any SIP special module or something like this in order to let SIP+RTP work OK ??? Special thanks Alejandro

Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot

Dear all, I have an Asterisk 1.4.13 with SIP protocol and several voip clients (Twinkle, X-Lite and SJPhone). Every call among voip clients pass through the Asterisk server, so there isn't any voip packet client-to-client. Can Asterisk control the RTP open ports the voip clients use ??? Or the RTP open ports depend on the voip clients ??? Special thanks Alejandro

Dear all, I have Asterisk 1.4.13 as a SIP server for my company with 100 peers (I mean users) and everything work fine. I have the following question: what is the maximum number of peers that I can reach with Asterisk ??? I know Asterisk is not a SIP server basically like OpenSER, so I'm confused. Thanks a lot, Alejandro

Dear people, does anybody try the ZRTP patch for Asterisk in order to have ZRTP encrytion among SIP/RTP calls ??? In other words, did anybody succesfully implement ZRTP in Asterisk ??? Any documentation about it ??? Special thanks Alejandro

Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with "envelope=yes" and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full: