similar to: Asterisk and OH323

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom

The configuration I'm building replaces an existing PBX with Asterisk. There are 8 existing modems that people use to call in from the outside to connect to PC(s) on the inside to transfer data, etc. Callers access these modems by calling the main number and then dialing an extension for the modem they want to talk to. What are my options for supporting these modems with Asterisk? Here are

Hello, i have some problems with compiling oh_323 (0.6.1) and asterisk (0.9). I successfully have compiled the necessary libs pwlib and openh323. I have set all path-variables in the oh_323 makefile. When i compile the oh_323 channel driver i'v got some errors. in function oh323_call chan_oh323.c: 1135 ... too few arguments to function 'ast_queue_hangup' So what could be the

Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the

Hello, I have modified the VoiceMailMain application to satisfy the request of the "local users", i.e., my wife. She lost patience with too many options (we have one mailbox, so we don't need to forward messages, or reply to messages, or file them in 6 different folders...) So the modified app says "Message 1", reads the message, "Message 2", reads...

According to the wiki at voip-info.org, the dial structure for using oh323 without a gatekeeper is: OH323/<exten>@<host>:<port> or OH323/<exten> The second option is valid only in the case where a gatekeeper is used. NOTE: OpenH323 library v1.12.0 has a bug in the parsing of the destination host. When this version is used then the above syntax should be:

Hi list, I've got a testbed running with the following config: 1. RH 7.3 linux machine 2. 2 Grandstream phones 3. 2 XTen soft clients When I dial voice mail I have a problem. here is the flow. 1. dial 8500 (the exten for voice mail) 2. enter 2600 via touch pad on the grandstreem 3. entire passwd via touch pad I always get an invalid login When looking at the Ast console I see that the

Hi, i'm trying to setup Asterisk to work with GnuGK using the Openh323 channel driver. I have a Gatekeeper that gets H.323 calls from a Cisco GW. To this Gatekeeper I've registered some endpoints, Cisco ATA186, Snom 100, etc. Now i want send the numbers 083xxx into Asterisk. Easy, i'll just enter something like this into oh323.conf: gwprefix=083 And all my calls starting with 083

Hello all. I am a little (allot) lost on my next hurdle in getting an asterisk system built. I would like to get my asterisk servers configured exclusively from database. I have read through the wiki on this subject but once again I find that there is a certain level of knowledge that is assumed. As of now I know nothing about databases in general and specifically MYSQL. I do not know

Hi, does anybody manage to get music-on-hold with inaccess oh323 driver? Statement like : exten => 10,1,Dial(OH323/xx,mt) works (dials the xx number) but no music is heared. Also, if I put 'r' (ringback) it doesn't work either. With chan_h323 I got this functionality but this driver had some other problems (call transfer don't work).... Thanx in advance, Victor...

Hello everybody, First time installer and I need the lists advice. My plan is to use asterisk PBX with some hardware to terminate my calls coming from several operational gnugk gatekeepers. Do have RH9 and downloaded the latest asterisk from CVS. Compiled according instructions and is running fine. Could hardly find any info on h323 implementation untill the REAME in the channels directory.

Hello all, Please forgive me if this sounds a little (or a lot) ignorant as I am very new to asterisk. Right now I have two pc's connected back to back through an E100 card running asterisk. I have openh323 running as well and I am able to route calls through the E1 line. Next up I would like to be able to register asterisk with a gatekeeper. On another computer is running openGK. Using

I have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - AVS@210.21.118.XXX Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn-xxxx e.164 - 22xx2912

Hi Hekuran I have installed asterisk-1.0-RC2.tar.gz, asterisk-prepaid-0.3.1 and postgresql. When I tried to call from any IAX client to another IAX client and also sip client to sip client it worked fine. And also the cdr table filled properly. Now I tried to configure asterisk-prepaid-0.3.1 with asterisk. I have compiled asterisk-prepaid-0.3.1 and also copy the configure file. I

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. " WARNING[8446]: chan_oh323.c:3555

Due to the format of the message coming from the H323 channels included w/ Asterisk we were unable to use our gatekeeper. For a quick solution we tried the OH323 channel drivers and can receive inbound calls from the parent gatekeeper. We are trying to do a dial to gatekeeper... I am trying exten => 5551212,1,Wait,2 exten => 5551212,2,Dial,OH323/5551212 But I am not sure if this is the

Hi: Is DynExtebDB module still working?? -- JO

I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323

Hi, I've oh323 chan installed and working to make calls from SIP to H323 devices. The problem is can no hear sound with the H323 device. I think this is some related with codecs o nat, because the H323 have one public IP from a different subnet from the asterisk box. If I use netmeeting in gateway mode, the call can be completed and I can talk with a SIP device, but in gateway mode I can not