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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.

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

> 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 outside the firewall with a registered IP address, the SIP phones

Hello! i'm trying to set up transfer without using the respective asterisk-function but with the built-in phone functions. my goal is to have the first callleg billed to the caller and the second callleg to the callee, who is responsible for the forward(and i can't bill a unknown caller anyways) so far it's working without problems, but my cdr's are messed. with the help of the

Has anyone tried installing * on a box with two eth interfaces which is acting as a NAT box? I have only one IP at this point and I would like to get * working without all of the NAT issues. My idea is to run * on my gateway (which is also running the firewall and masquerade services). All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the NAT screen, and will connect to the *

Hi All, When the Caller hangup at the voice menu, the wildcard X100P didn't disconnect the calls properly and it just keep looping at the voice menu and timeout and loop again, are there any methods can fix the problems? Please help! Thanks, Randal

*This message was transferred with a trial version of CommuniGate(tm) Pro* Will asterisk actually convert between two different codecs????? ie, a SIP endpoint running GSM and another running G.711? Wouldn't that add quite some latency? I was always under the impression Asterisk did not recompress and was smart enough to negotiate the right codec at each end and just pass through the RTP

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Hi all, How do I route calls based on the DID the incoming caller dials? I?d like someone calling a DID to by-pass the main menu prompt and dial the extension associated with that DID directly. Thanks Michael Rose, Jr. ? ? ?

Hello everyone. I am having problems getting speex support. It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install

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. > >

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Hello, I've successfully installed Asterisk from last CVS and configured it for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip server. All are work fine at G711 codecs, but then I disable all codecs except g729 some calls failed (Not all calls. Some calls passed at g729 succesfully). All my devices configred to use only g729 and I don't see other codecs at mgcp or sip

Hello list, I am glad to announce that XC-AST version 0.9.0 is out today. New functionalities include: * Though not yet available to the end user, this release inclued the basis of the Outbounds Call Manager that will be released for 1.0. If you update from a previous version, have a look at the UPDATING.txt to understand how to upgrade your database schema. * The realtime visualization

Is there a way i can change my subcription email address, without unsbubbing and resubbing myself? Cheers, Mathew -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20030221/c9149135/attachment.html>

Hi, My system is like this currently: ATA-186 <-> *1 <-> IAX2 to Europe <-> *2 <-> i4l <-> voicemail at cell provider When I dial up to my voicemail at my European cell phone provider I can't press '#' to get into their menu. It seems like it just ignores any DTMF tones or doesn't get them. When I call a human on the other side of the i4l they

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

I have this configuration: UA1 ---> FW1 ---> Asterisk ----> FW2 --> Internet --> UA2 UA has provate address (192.168.x.x) Asterisk has public address I want to be reach somebody at the internet. My idea was that asterisk works as a Proxy. Then i would have a SIP/RTP connection between UA1 and Asterisk and an other SIP/RTP connection between Asterisk and UA2. (asterisk is

Trying to configure voicemail with H.323 all I get is the following errors when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it might be a simple fix. [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by inAccess Networks