similar to: chan_oh323 and DTMF

Hi, all!!! What will Asterisk do in the following case: For example, we have 4 licenses, and have 4 simultaneous calls, using G729. Will asterisk allow incoming calls from peer, that can talk G729 and ulaw, and will it force it somehow to use ulaw in this case? All phones there in LAN behind Asterisk prefer GSM codec, so it does transcoding. So, what I mean is will Asterisk fall back to use

I've successfully got inbound/outbound calling working with our Asterisk using the Asterisk-OH323 channel driver. We are using a parent gatekeeper and the NuFone H323 channel driver would not work with the parent gatekeeper... I'm trying to determine a way to ensure that the line used for outbound calling is always available i.e. like trunking.. >From what I can tell when I place an

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

Hi, We've got endpoints and gateways who have T.38 fax support. We now use SER and Asterisk to do our routing and other functionality, but fax doesn't seem to work. Asterisk complains like this: Sep 9 09:25:45 WARNING[467828746]: RTP Read too short Sep 9 09:25:45 WARNING[467828746]: Asked to transmit frame type 256, while native formats is 32 (read/write = 256/256) With lots of the

asterisk-users-bounces@lists.digium.com wrote: > Isn't it possible to use T.38 for interconnecting hardware gates > supporting T.38 with asterisk using SIP REINVITE? > I'm not shure but but think its's might be possible because after > reinvite traffic goes directly from one gate to anotger, not over > Asterisk We've seen a problem here with asterisk. Wehn

I'm just curious if anyone has the X100P & Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin

Hi, need a quick help ... it should be easy but ... exten =>_9898,1,Answer exten =>_9898,2,VoiceMailMain(${CALLERID}@domain) Accepting overlap call from '342' to '9' on channel 0/2, span 3 -- Executing Answer("Zap/8-1", "") in new stack -- Executing VoiceMailMain("Zap/8-1", "@domain") in new stack As you can see there

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hi, I run asterisk with oh323 plugins.It runs correctly with sjphone H323 Gatekeeper. But When i run openphone it doesn't recognize my asterisk server like a gatekeeper !! What is the problem ? Thx

I am looking at building an IVR product with a few interesting features and need some more information about how asterisk and VoIP work and what I can get from them. As far as I can tell when I use ISDN/GSM telephone networks the DTMF information travels as data representing 'start tone' and 'stop tone' for each button pressed, it is then generated at the other end if an

Hello, I have installed oh323 channel driver. Outgoing calls to H.323 world do not include RFC2833 in the outgoing TerminalCapabilitiesSet despite that userInputMode=RFC2833 has already been set. Does anyone know how to make RFC 2833 DTMF relay work over oh323 channel? Kind regards, Fernando Herrera _____ De: Fernando Herrera [mailto:fherrera@iplan.com.ar] Enviado el:

Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")

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 there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

Hello, I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5). Everything is working fine, well, except that : when a call is made from an h323 device (gnomemeeting for example), the caller does not hear any ringing at all, he suddenly hears the person who answers the phone. That can be quite disturbing for the users. Any help would be very welcome. thank you. Yves

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

Hi all, Sorry if this has been answered previously, but I have not had any luck trying to find it. I am trying to connect my Asterisk server (1.0 stable, Fedora Core 2, kernel 2.6.8-1.521) to connect to a gateway that can only support H323. I have installed the asterisk-oh323 channel driver (version 0.6.3b) using Open H323 1.13.5 (patched as per asterisk-oh323's instructions) and PWLIB

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, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same