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Hi List, I am sorry that I may bring the old question to the community. My question is 1. How can we determine if asterisk is working normally or not ? what kind watchdog process do we have at this moment ? 2. In case the running asterisk is mulfucntion, is there any available way to auto restart asterisk ?? Please advise if you could. Thanks.

Just in case other people on the list have this problem... Begin forwarded message: > From: "George Lin" <glin@cosini.com> > Date: Thu Aug 14, 2003 6:54:46 AM Europe/Budapest > To: "Paul Cheng" <asterisk@klarium.com> > Subject: RE: FW: [Asterisk-Users] SIP NAT question > > Dear Paul, > > Thanks for the suggestion. It works now. > > Thank you very much. > &g...

> From: George Lin [mailto:glin@cosini.com] > > I want to deploy multiple SIPs phone in our office. And we have shutdown > the > firewall at our office router(with ip 211.x.x.x). we have deployed the > asterisk with IP 218.x.x.x. > > All SIP phones have 192.x.x.x. We have something similar George, * sits o...

Hi all, I'm setting up my first * install and have it peering with another * machine using IAX across the internet which provides our pstn gateway. So far I have the IAX "friend" set up correctly but when I make a test call from an external phone, I get: WARNING[5126]: File chan_iax.c, Line 648 (get_timelen): Don't know how to calculate timelen on 8 packets I have set up a

Hi List I have one question about MGCP in asterisk. I have a media gateway, and I want to have asterisk to work with the media gateway. As I was told that the media gateway can communicate with the switch via standard interface MGCP/ICGP. Question is if the asterisk MGCP supports such MGCP message ??? Thanks. George Lin

Hi guys,

...n anyone give me a direction to solve this problem ? Thanks in advance, Areski _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 5 From: "G Lin" <glin@cosini.com> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] H.323 and G729: Another sad tale Date: Thu, 6 Nov 2003 08:33:09 -0800 Reply-To: asterisk-users@lists.digium.com Dear all, I just fresh CVS the asterisk code, and uncomment the G729 in the Makefile on asteris...

Hello, Can someone tell us if current asterisk supports FAX over MGCP, SIP and H323 ?? Thanks, George Lin

Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind

> Hello all, > Can someone help me on the problem which I have on MGCP phone test . I test mgcp - asterisk- zap. But I got several NOTICE message from rtp.c. > NOTICE[20501]: File rtp.c, Line 221 (process_rfc3389): RFC3389 support > incomplete. Turn off on client if possible > > -- Endpoint 'aaln/1@VG101-1-1' observed '9' > NOTICE[20501]: File rtp.c,