similar to: Asteriks, GnuGk and outgoing calls

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

Help, Any ideas ? I guess I missing something. make[1]: Entering directory `/usr/src/asterisk/utils' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\"CVS-HEAD-12/27/04-21:28:39\" -DASTERISK_VERSION_NUM =999999 -DINSTALL_PREFIX=\"\"

I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with asteriks@home ver 0.6? Little has been mentioned about use of quicknet products' adaptability with asteriks@home I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __________________________________ Celebrate Yahoo!'s 10th

We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on "attended transfers". What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to

How can I monitor asterisk if all lines are registered etc? I have an asterisk on a remote location and sometime they reporting problems that phone is not ringing, they can not dial out etc. Usually I just restart asterisk and it solves the problem. Is there an application that will email me if case any line looses registration with with asterisk? Or any better solution! -- Joseph

Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue

Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it --------------------------------- Get your own web address. Have a HUGE year through Yahoo! Small Business. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I was having question about mysql driver support ( not odbc). Do we still need the asterisk-add-on to be installed for mysql support.? If yes, Which version should be used and from where I should get it? Thanks in adavance. ---- Thanks & Regards, PrashantAbhang -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -------------- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad

Hello All. I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience. I want to use asterisk for call park/pickup and have configured openser to relay calls made to ruri 700-720 to asterisk running on localhost:5069 Call flow: phone A calls phone B (both phones are polycom) Phone B answers then phone b

Dear Alex; Thanks for your kindly reply. Please explain for me what do u mean exactly in "a la" in the following sentence u wrote it below? " in SIP, this can be done via "re-INVITEs" a la the canreinvite= option for SIP peers in sip.conf" Another thing, do u mean that it is easier (better) if we need H.323 endpoint to talk with SIP endpoint then we use full

Hello * users, I have a problem with authenticating my SIP gateway endpoints with *. The gateway I'm using is an AudioCodes MP-124 (24 port) If I setup my sip.conf with an empty secret= option everything is working ok and I can initiate and receive calls, but if I want to be able to authenticate my SIP gateway with * something goes wrong. Below is a little snip of my sip.conf file: ---

Hi, We have got a new PRI card at one of our Office locations and now I need to install the the device on a remote server. Is there any way to know if the device is loaded already. When I give " cat /proc/zaptel/* " it returns the following. # cat /proc/zaptel/* Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER) B8ZS/ESF RED IRQ misses: 2 1 WCT1/0/1

Isn't possible to test the ntlm_auth with the ntlmssp protocol in a command line mode, you must use a browser able to handle ntlm because only this sort of browser send the appropriate ntlm challenges, try IE. Estevam Henrique -----Mensagem original----- De: samba-bounces+ecarvalho=bmf.com.br@lists.samba.org [mailto:samba-bounces+ecarvalho=bmf.com.br@lists.samba.org] Em nome de Champaka