similar to: h323 how to set it up?

Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =

My nightly logwatch report had a never before seen section last night, "barracuda spam firewall". I have not problem with the emails it noted as being rejected. But I've always thought of "barracuda" as a commercial product. I have neither configured nor enabled any barracuda software and "yum list '*barrac*'" comes up empty. What is this? Jon -- Jon

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0", "H323/652#150388590962@1.1.1.1|60") in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28

Is this list archived anywhere in an easily searchable format? I'd like to check the archives before asking a question that's already been answered... tia, Christian ---------------------------------------------- Christian Cryder [christianc@atmreports.com] Internet Architect, ATMReports.com Barracuda - http://barracuda.enhydra.org ----------------------------------------------

Roland, On 2020-09-10 21:27, Roland wrote: >> with rsync hanging - after breakout on /home for writing I then get: >> "Read-only file system" > > if your filesystem switches to read-only, you have a serious problem > with your system/storage, not with rsync. > > rsync (or the workload) is simply triggering the problem. Thanks for the response . . Hmm . .

Hi, I had a power failure, and the on-disk configuration for vinum went bizarre. The logs read from disks are at http://biaix.org/pk/debug/ (log.$DEVICE files). The logs in da0 (barracuda) are the ones obviously wrong, I'm pretty sure the others are ok. Is this a 'virtually' dead drive? Can I force vinum to use the other's drive configuration? What's the less traumatic

I'm really struggling with an Asterisk 1.6.2.7 install (on centos 5.4) The pwlib + opal packages don't satisfy Asterisk's configure script (to let H323 compile), so I removed those and added the latest ptlib + h323plus (from h323plus.org) I can compile ptlib and h323, but when I load chan_h323 in asterisk I get a segfault. I had to point LD_LIBRARY_PATH to /usr/local/lib with the

Hi, I use with asterisk gnugk a gatekeeper for h323 client. I don't understand why asterisk can't have the H323-ID (callerID). In the gatekeeper's monitor I have this H323-ID but not in asterisk. Does anyone know something about it, or how can I send a caller ID to asterisk ? Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

The man page does not say much, but does this mean I have only 396668 used by programs (used-cached)? Or shoul I be reading the 2nd line? [root at ten-212 ~]# free total used free shared buffers cached Mem: 7918844 5478820 2440024 0 111684 5082152 -/+ buffers/cache: 284984 7633860 Swap: 9961464 204 9961260 --

I receive the following error when I try to call another H323 extension from another H323 when going through *. NOTICE[27669]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 8 NOTICE[27669]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 8 to 1 WARNING[27669]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 1,

Hi all. I want to use two servers with dovecot using a common file system with drbd. So I have several questions. If one server write a mail to th file system he will use his name as part of the mail identification. the second server will use his own name. Each server will generate it own mail numbers . When an imap or pop user will consult the mails i shoul be confusion. Am I correct. is there

Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more,

Hi, what's the difference between h323 and oh323 modules? which one should I use? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030722/3d3edb73/attachment.htm

I am trying to calculate the regression for the follwing input data stored in 'data.txt' file.I am reading this and storing it in the variable i .then i am trying to get the predicted value using f1 as dependent and others f2....f10 as independent variables.It is giving the following error. Also i want that i shoul get one predicted value for each row(y). What should i do. Please help me

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

Hi, Someone released a succefull interconnection in H323 with WTL equipement ? I'm trying to do that with an IPNx. But get dead air. With chan_oh323 it's fine, all works. With chan_h323 => dead air. The configuration is GW to GW. This is my configuration from h323.conf: [general] port=1720 bindaddr=my.ipaddr dtmfmode=rfc2833

Hi everyone, Could someone help me on make my Asterisk registers to a Gatekeeper. I have compiled the chan_h323.so and it seems to be working. What I want to know is how can I "route" my SIP clients to a single account on a remote Gatekeeper. I have tried a lot of conbinations but nothing happend. For example: my account number: 123456789 ;extension.conf exten =>

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.