similar to: sip <-> oh323 / real-time / g729 - one way audio

Hi, pwlib 1.6.6 & downloaded & ./configure & make it as written The same with openh323-1.13.5 Downloaded & patched make & ./configure & make it as written Then with asterisk-oh323-0.7.1 Downloaded (I used u file there to patch openh323) Made some changes in the Makefile to adjust directories Then 'make' I got an error in chan_oh323 : asterisk/channel_pvt.h

Hi, I have the same problem with cvs head. (1.0.6) See http://www.inaccessnetworks.com/projects/asterisk-oh323 And https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php (issue 00...008) some 'patch' files are included. I am a newbie to linux and asterisk. I do not want to blow my config. Please give me a feed-back if those files helped you and how. Also if you have a

Well here's a suggestion - a little crazy - but works... Most equipment is taking the 120vac and converting it into DC voltage. So why not just feed it DC voltage directly??? We had a situation where our field techs needed to test dsl circuits and voip ata from the demarcation point outside a house or business. A UPS might have worked - but the down conversion of 12v dc battery in ups up to

Hi, cisco's phones are VoIP only polycom build (video-) conferencing devices. One Cisco model (7930 I thnk) is a polycom in disguise. The code is not 'cisco-like' (at least the version I had. Both brands make very good quality equipments. Good sound, good support, ... Regards, Shaoul Jacobson VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail

We have one Swissvoice IP10S running SIP firmware. Recently, I've been getting these messages: Mar 1 13:59:44 NOTICE[20933]: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Any clues off the bat? I'm still researching other stuff.. Thanks, Matthew

Hi list! Sorry for this slightly off-topic message but does anybody know if the standard for ISDN BRI is the same in Spain as it is in the rest of Europe (or the Netherlands). Will a standard HFC-S card work?

Hello: Has anyone on this list had to configure hairpinning on a Cisco gateway running IOS 12.2 or 12.3 and using a PRI for connectivity to the PSTN? If so could you tell me how it is done? I'm told this is the source of my call transfer problems and yet I cannot find clear instructions for how the configuration is done. Thanks,Steve -- ISC Network Engineering The University of

Dear Fellow *-ers, First, you guys are fantastic. Keep fighting the good fight. Second, it sounds like comments in the code are coming, which sounds welcome by all, even those of us who couldn't code their way out of a papersack, but who need to read the source. Last, I might be traveling to Europe (from US) & want to tow along hardware & haven't done this before & was

I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco

Hi..... I am new to this software, and I want to implement a client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not

Good day, I am trying to setup an outbound dial plan which will time out if no answer. Using a X100P with the following dial command : exten => 101,3,Dial(Zap/1/3036972357,5) ; try the desk line - fail to step 104 It dials out successfully, but never times out. I have a basic Zapata config : group = 1 context = RedRockWeb language = en signalling = fxs_ks usecallerid = yes hidecallerid

Is uniqueid globally unique? I have three Asterisk installations and I need to store data from all of them in same database, in same table. Will this uniqueid field be unique? -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr -------------- next part -------------- A

Hi Jim, Thanks a lot for investigating. It definitely makes sense now. I'll fix the problem now. Is there any other place where you see that same (or similar) problem? Jean-Marc Jim Crichton a ?crit : > Jean-Marc, > > I dug into this further, and found that the warning occurred when PSHR32 > had a shift greater than 15. > > in fixed_generic.h, PSHR32 is defined as: >

Thanks Jim for looking into that, I was really starting to wonder what was going on. Let me know if you find a way to tell the compiler to stop complaining. Jean-Marc Jim Crichton a ?crit : > I looked back at my old C55 EC build, and I had the same warning in > mdf.c which Mike found. The assembly code did have a valid shift, and > this build did cancel echo. > > When I built

Dear All, I have problem with new oh323 0.6.3a , I try use var OH323_OUTCODEC, but it don't work. oh323 driver don't want connect to gateway with g729, it's work if I only use in oh323.conf one codec ( g729 ). If I enable 2 or more codecs - always in use other codec: -- Executing SetVar("IAX2[4010@4010]/1", "OH323_OUTCODEC=g729a") in new stack -- Executing

Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb ram, with g729 for i686 , (fedora 1). my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen otherparty realtime voice , but other party geting sip party's voice 1 sec later (not

Hello list, I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider I have this problem: oh323 (last version): ------------- asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable

Hello list, I have many IP hardphones like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider ( G729 from digium ) I have this problems: oh323 (last version): ------------- asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider (

Hi everyone, Could somebody enlighten me on this one? I have configured my asterisk to run on oh323 using codec g729. Incoming calls are working okay. But the thing I want to work is say pressing some options, say dial 1 to go to voicemail or dial a certain number to dial a specific extension. I have a config for this and tried calling from a normal PSTN and is working. But i just can't seem

Jim Crichton a ?crit : > I played briefly with the echo canceller on the C5509A back in May > 2006. I got the same compiler warnings, and sent a message to the > list which included this: > > "I got several compiler warnings for "shift out of range" in mdf.c, > which I fixed by adding EXTEND32 to all of the SHL32s with 16 bit > operands (st->frame_size in 6