similar to: sip to h323 ... direct RTP?

Hi, I am familiar with asterisk, though never actually tinkered with one myself ... so i don't know the full extent of its capabilities. I am facing a request to bridge a sip network and an h323 network. I would like to operate the sip with ser as the proxy and some gatekeeper on the h323 side (not required though). Actually, i have a few more points that may make it simpler - i do not need

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Friends in the Asterisk community, I've been talking for years about the new version of the SIP channel. I've been trying to get funding and get going. Well, the funding part remains to be handled, but I have other news - if you kan keep it to yourself. ...I've began coding. Finally. With a happy smile on my face I removed "pedantic=yes" the other day. After years of

Hello, I am altering an asterisk configuration and would like to eliminate the loading of modules I do not want or do not need at the moment. For example I am do not want to use chan_zap (I'm using chan_capi) and don't want to be bothered with music on hold at the moment. Is there a way to configure these things off so asterisk doesn't try to load them? Or do I have to just

Friends, Beginning next week, I will travel around Europe to teach Asterisk - the one day Meet Asterisk training. MeetAsterisk is organized by Edvina in cooperation with Digium and Voop. In many places, local Asterisk equipment resellers participate and show their equipment. This is the tour plan: * Amsterdam April 26 * Copenhagen April 27 * Oslo April 28 * Paris May 3 * Brussels May 4 *

Hi, I would need to bridge a SIP call with a multicast RTP channel. Both sides are receiving and transmitting RTP. Googling, I saw that an app_rtppage, which was in the SVN for a while and its not there anymore. It did, I think, only partly what I need (it sent from SIP to the mcast ... not the other way around), but it was a start. Any idea how to do this? I also could use

Hi, I need an Asterisk with G729 support. Preference is with Asterisk 1.2(.18), but if not possible, then it can be 1.4. Question is, can I enable G729 for both protocols? do the H323 implementation allow it? I found the codec support for H323 in 1.2.18 very poor ... only got u/a-law to work ... not even GSM. Would the Digium G729 license be good both for SIP and H323? Cesc

Hi, I know it is a bit off-topic, but I'd like to ask the community what is the current most supported way to deal with DTMF? I'm looking for an all-SIP system and I'm mostly interested in the end devices support of the different methods (DTMF in-band audio, DTMF RTP telephony events packets, SIP INFO, ...) Thanks in advance. Cesc -------------- next part -------------- An HTML

Hi, I have asterisk 1.2.18. I just took a peak at the command: > show translation and I saw that I can only convert from/to ulaw, ulaw, gsm and slin. No speex, no ilbc ... do I need a license or compile something extra? The G723, 726 and 729 ... I need a license, is that it? one for all of them? or for each? How do I get them to work? not just pass-through ... I need conversion. Thanks a

Hi Just uploaded source from cvs (CVS-HEAD-11/27/04-12:56:51) Zaptel and libpri make install works ok, but I get the following error when running make install in asterisk directory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ........... make *** [rpt.o] : Error 1 What have I done wrong ? (Its got to

Thanks in advance for your help. I am running a repeated measures ANOVA in r. The same group undergoes to four different treatment conditions. So, all individuals are treated with treatments A, B, C and D in four different occasions. Once I get a significant ANOVA, I first run a paired samples t-test using the code: t.test(X1,X2,paired=TRUE) #being x1 the punctuation after treatment 1 and x2 the

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

Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM]

Greetings: I have a working configuration for SCCP on our LANS which doesn't route RTP correctly to a skinny phone behind NAT registering from a remote public IP. Configuration: asterisk 1.4.35 servicing only skinny phones trunked to asterisk 1.2.40 which services chan_phone FXS, zap FXO and SIP phones; both instances of asterisk are behind NAT and run on the same host (using different base

Hi, I hope this gets picked up by some bug marshall ... I have downloaded (yesterday) the 1.2 branch from svn ... When running: asterisk -vvvvc loaded modules: [modules] autoload=no load => pbx_functions.so load => pbx_config.so load => codec_a_mu.so load => format_pcm_alaw.so load => codec_ulaw.so load => codec_alaw.so load => format_pcm.so load => func_uri.so

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

It seems renouveau doesn't cope well with missing GLX extension, unlike e.g. glxgears: rmh at cesc:~/renouveau$ glxgears Xlib: extension "GLX" missing on display ":0.0". Error: couldn't get an RGB, Double-buffered visual rmh at cesc:~/renouveau$ ./renouveau detect_devices: Creating probe window failed. We tried to create a window by using SDL. Our OpenGL tests require

I was able to get app_prepaid working, but unfortunately I am getting one way audio on the phone that I was placing the call from. It is behind NAT. It appears that the app_prepaid is not taking this into consideration since I see: Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route: build_route: Contact hop: <sip:7708183799@192.168.1.101:5060;line=jet7pbic> Jun 18 17:46:25

A friend of mine just sent me this email - he is looking for an IAX inbound service in Holland - any thoughts? Voip info only has Nadiz which seems to be SIP only. Hi Dean, I need a Dutch number with IAX support. Do you have any leads in that direction? It's been difficult for me to figure it out -- especially since most of their sites seem to be in Dutch... Regards,

The Asterisk Developer Team is proud to announce the Asterisk SPE v1.0 Beta Release for immediate download on tftp.digium.com. The SPE has been developed as a joint project between Digium, the Asterisk Company, Voop, the European Asterisk Dialtone provider and the Asterisk community. The Asterisk Service Provider Edition is focused on the needs for the new breed of Telecom companies - the