similar to: Problem with MeetMe Conference!!!

Displaying 20 results from an estimated 800 matches similar to: "Problem with MeetMe Conference!!!"

2006 Mar 03
1
Problem with NAT!!!
Hi all I'm a newbie in asterisk.I install asterisk server successfully. I configure this server to traverse NAT. Using Xlite clients, i make a call between 2 local networks through Internet.Asterisk server is installed on a host with public IP. client A (in the LAn A) and client B (in the LAN B) are registered. When i make a call from the LAN A to the LAn B, everything goes well.But, when
2006 Mar 17
1
french sounds in asterisk
Hi all i want to know where i can find french sounds for asterisk. I don't have any studio to register good sounds. Bests regards Serge ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et
2006 Jan 30
1
app_snmp
Hello, Is there an app_snmp for asterisk-1.2.3 ? Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com
2006 Mar 04
0
RE: Asterisk-Users Digest, Vol 20, Issue 20
Message: 6 Date: Fri, 03 Mar 2006 17:32:47 +0000 From: Conrad Wood <asterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.net> Subject: Re: [Asterisk-Users] Problem with NAT!!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:
2006 Mar 07
3
Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?
Hello, I use both ser/asterisk . In fact i wish asterisk to forward all the sip requests which are not handled by domain=domain.tld in sip.conf Here is a diagram: The sip agents use the Sip proxy as an outbound sip proxy and domain=domain.tld . When the sip agents dial sip:user@otherdomain.tld so the request is sent to sip proxy and so to Asterisk. I wish Asterisk to Look up the
2006 Apr 07
2
407 proxy authentication
Hello, Asterisk sent back 407 proxy authentication . How can avoid this ? I set insecure=very without success in sip.conf and my sql server . Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur
2006 Apr 08
2
HELP !!!!!
Hello, I wish to set a sip uri sip:info@mydomain. I use ser for authorization and authentication (registrar rtpproxy and outbound proxy) I use asterisk 1.2.5 with realtime . the info is used as a hunt group so i add in extension.conf [info] exten => info,1,Answer() exten => info,n,Dial(Sip/84,10) exten => info,n,Dial(Sip/85,10) exten => info,n,Hangup Ser forward sip:info@mydomain
2006 Jan 30
3
How many digium cards per server ?
Hello, How many digium cards is supported per asterisk server ? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com
2006 Jan 29
1
file.c:509 ast_openstream_full: File 100 does not exist in any format
Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory Regards
2006 Mar 02
0
Redirect a sip outbound requests to a sip proxy
Hi all, Is there a solution to solve this ? ASTERISK 1.2.4 || Internet===SER/OPENSER=====Nat==[private net] || sip agents rtpproxy/mediaproxy Sip agents use SER/OPENSER as an outbound sip proxy and asterisk as a registar server, pbx functions, ... SER/OPENSER look for domains in URI. if domains are handled by SER/OPENSER
2006 Mar 05
0
to configure asterisk to work with the nathelper module of openser
Hi all I'm a newbie in asterisk.I ant to know how i ca configure asterisk to work with the nathelper module of openser to fix the nat problem! Thanks in advance! bets regards Serge --------------------------------- Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international.T?l?chargez
2006 Mar 06
0
Outbound Proxy Support
Hi all, May I have to patch asterisk-1.2.x with this patch http://bugs.digium.com/bug_view_page.php?bug_id=0002859 to configure an outbound sip proxy in sip.conf ? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et
2006 Mar 21
0
app_queue and ARA
Hello, I've configured ACD with ARA asterisk-1.2.4 . I try "show queues" command but no queue is shown. why ? Can I keep the caller on queue until an agent answer the call ? I use ARA to configure queues and members however i have to use agents.conf to store the agents. I wish to configure agents in SQL db. Is it possible ? Regards Harry
2006 Mar 22
0
TIMEOUT(s)
Hello, Here is part of my extensions.conf. I set both absolute and response timeouts according to the "day" context. I wish to asterisk hangup after 60s and 10s to play or replay the annoucement . Asterisk doesn't jump to T extension. How can fiox this problem ? harry ... [day] exten => s,1,Set(TIMEOUT(absolute) = 60) exten => s,2,Set(TIMEOUT(response) = 10) exten =>
2006 Mar 30
0
Setting up announcement on reply to 4xx 5xx 6xx messages
Hello, I wish to play a recorded announcement on reply to 4xx 5xx 6xx messages . According to the status a audio file would be played from asterisk server via ser to the caller How can I configure a such feature ? My configuration: Ser act as an outbound sip proxy . Asterisk a sip media server and registrar. sip agents ---- SER -----Asterisk Harry
2006 Mar 31
0
No voice heard in festivalassociated with asterisk!!!
Hi all I complile asterisk 1.2.4 successfully.I install festival successfully and i configure asterisk to work with festival.But When i call the festival extension configured in extensions, the festival application is executed well (i see it in the log) and must read the text (hello world).But i'm hear no voice. What's the problem? Thanks
2006 Mar 31
0
decrease the speed of reading text!!!
Hi all How can i decrease the speed of festival? It appear that in festival, the text is read too fast for me ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com
2006 Apr 10
1
Call me for testing my system
Dear User, Anybody could dial these sip uri : sip:info@nxs.yi.org (french) sip:music@nxs.yi.org (music 60s) sip:support@nxs.yi.org (french) Thanks for help ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur
2006 Apr 08
0
Re: [asterisk-dev] bug or bad chan_sip.c
Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes
2006 Mar 09
2
TDM11B Hang up detection not working in France ?
Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,