search for: ronald_wiplinger

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

...Call Transfer > using SIP clients (Frank Schoep) > 3. RE: presence and IM again, want to develop a > working"hack" > (Florian Overkamp) > 4. calling shell scripts from within * (Terry > Wade) > 5. Re: Sometimes yes - sometimes no (dialplan) > (Ronald_Wiplinger) > 6. Dialogic D/300 E1 (Fredrik Lith?n) > 7. Transfer and CDR's (Sebastian Zaprzalski) > 8. RE: Provider Survey (Mohamed Farid) > 9. oh323 problem with cisco 2600 (craz sead) > 10. Re: [SPAM:***** SpamScore] Re: > [Asterisk-Users] Call Transfer > using...

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic

I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. What could be the reason?

Is there a possibility to send text based messages from/to a sip phone (text display) or to a video phone or from/to a messenger? bye Ronald

Matthew, can we use real-time also for H.323 phones? (h323_buddies) ??? bye Ronald

...teliax down? (Jay Milk) > 15. RE: LiveVoip is Bankrupt (Jay Milk) > 16. RE: Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE! > (harry gaillac) > 17. RE: LiveVoip is Bankrupt (Terry H. Gilsenan) > 18. Can anyone guide me regarding h323.cong ??? > (Adeel -31) > 19. H323 (Ronald_Wiplinger) > 20. Re: SixTel? (Erik Espinoza) > 21. Shoutcast Music On Hold problems? (hank) > 22. Re: Eicon equipment, BRI Server or PRI? (Armin > Schindler) > 23. Re: polycom soundpoint ip 300 (Wilson Pickett) > 24. RE: RTP session between two end users (Erdem > HAK?) > 2...

I want to use Asterisk registering itself to a GK. SIP phones are registered to Asterisk H323 are registered to the GK I want to: 1. make calls from SIP (Asterisk) <--> H323 (GK) 2. use Meetme to make a conference call for both types of phones I got on the GK, login and password, IP of GK, and codex g711u How to set-up h323.conf and extensions.conf for that? (I am using ooh323c)

I am not aware what I have done wrong, but the result is a query of: *Database error:* Invalid SQL: SELECT * FROM WHERE UNIX_TIMESTAMP(calldate) >= UNIX_TIMESTAMP('2005-07-01') ORDER BY calldate DESC LIMIT 0,25 *MySQL Error*: 1064 (You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'WHERE

[public_gk] ;exten => _070.,1,Set(CALLERID(number)=070333333${CALLERIDNUM}) exten => _070.,1,Dial(H323/${EXTEN}@59.120.139.119) exten => _070.,n,Hangup *CLI> h323 show peers Name Accountcode ip:port Formats 7000 ast_h323 203.160.252.147:1720 0x4 (ulaw) 88670333333 ast_h323 203.160.252.147:1720 0x4 (ulaw)

Does anybody has a php code for using DBput (DBget, DBdel) from a web interface, which database is used for astrisk? bye Ronald

We found at the wiki a link to soekris and wonder if it is good? Is anybody using it and can share some experience, please? We would like to use it as a small PBX including a wireless access point, so that we can also use WiFi phones. bye Ronald

I want to make sure that RTP is not going thru my asterisk. I read you should avoid in the dial commands: m music while ringing t,T transfer calls from caller and called party What else do I need to take care? remote phone ===> registered to local asterisk ===> calling remote gateway should have the RTP remote phone ===(RTP)==> calling remote gateway bye Ronald

I don't see anything with sip show users and sip show peers, however it works! Is there a trick? I have installed realtime (sipbuddies) on one machine and see sip show peers/users and on my new installed system I don't. Have I forgotten something? bye Ronald

Can anybody spot the problem? H.323 via GK calls Asterisk box and should be connected directly to the extension 6002 (an voice sip phone) The caller hears only ringing. The called party hears the ring. Called party picks up, caller hears still the ring tone. Called party hears nothing. Called party hangs up, caller hears the busy tone. extensions.conf: exten => 88670333333,1,Wait(1) exten

I try to get H323 to run, but have so far only partial success: There is a Gatekeeper GK, where asterisk connects to. The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper. From the Network on the GK, asterisk is reachable via the number 070333333. I have an extension on asterisk 6002, which is reachable. I try to call a number attached to the gatekeeper (070168177) with the