similar to: Codec G729 / x86_64 bits.

Hello All, I'm facing problems with Unicall on this scenario : CentOS 4.3 - Running on x86_64 Asterisk 1.2.7.1 Zaptel 1.2.5 When running zttool , shows all Spans OK. But I can't receive and make calls. I tried to change many parameters and still doesn't work. Any clues ? * unicall.conf [channels] language=br context=incoming-pstn usecallerid=yes hidecallerid=no immediate=no

Is there any winblows softphone available offering g729 *and* IAX? I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones The best choice should be dIAX, but it is only GSM.

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

Hello list, I've posted my problem on BSD list and i still have the problem. The remote side receives the call , but there's no voice on the call. I tried everything about possible NAT problems .. but ther're on same net. My platform: FreeBSD 5.2.1-Release Asterisk 1.0-RC2 soft phones : X-Lite >>>> -- Executing Dial("SIP/1260-a7ae", "SIP/1262|20")

Hello All , In some ocasions i?m getting a problem with my X100P board. I?m trying to trace tre problem , but i didn?t find a possible answer. -> I get those messages when trying to use Zap Channel Sep 29 14:15:46 WARNING[-1094796368]: chan_sip.c:2107 sip_new: Unable to allocate channel structure Sep 29 14:15:46 NOTICE[-1094796368]: chan_sip.c:7283 handle_request: Unable to create/find

I'm trying to place a call from my Cisco 7960 and I'm receiving this error: Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel I can't place calls, but I can receive them: mail*CLI> sip show channels Peer User/ANR Call ID Seq

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

Hello list , I?m putting to work a new asterisk box. I?m running * 1.0.3 with (one ) Wildcard TDM ( 2FXS* 2FXO) and (one) E-100P. Both boards are working well. ztcfg don?t show me any error. zttool list both cards as "OK" But when i run asterisk on verbose mode i get those errors : [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de M?xico

Hello list, Is there some parameter on sip.conf to always let the client reachable ? I'm trying to avoid .... this situation : Sep 3 09:49:29 NOTICE[135442432]: chan_sip.c:7653 sip_poke_noanswer: Peer '1264' is now UNREACHABLE! Sep 3 09:49:39 NOTICE[135442432]: chan_sip.c:6408 handle_response: Peer '1264' is now REACHABLE! Regards, -Jefferson Carvalho

I'm trying to solve a mystery for the last couple of days. I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS. For several years, everything was working fine, no issues. A few days ago I started having problems at one particular site. NO CHANGES have been made to this office network - same router, switch and internet provider. No new equipment added or

Hi, I have installed AreskiCC on Slackware 10.1 with Asterisk latest CVS and Postgres 7.4. First of all the instructions are very confusing and hard to follow if you are not an expert. But, I managed to install it and.obviously t doesn't work. The other instructions I found on wiki are a great effort but incomplete. Basically the first thing that happens is that when I load

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I'm trying to add 2 FXO cards but when the second is added, asterisk stop to respond (zap channel): app_dial.c:545 dial_exec: Unable to create channel of type 'Zap' modprobe zaptel and modprobe wcfxo didn't return any error. ztcfg -vv is reporting only 1 card: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels

Most of my outbound calls are handled by macro contexts: [macro-pstn] exten => s,1,Wait(.5) exten => s,2,PlayBack(beep) exten => s,3,PlayBack(silence/1) exten => s,4,Dial(${ARG1}/${ARG2}${ARG3}) exten => s,5,Playtones(Congestion) exten => s,6,Wait(3) exten => s,7,Hangup which is called like exten => _920[1289]XXXX,1,Macro(pstn,Zap/g1,9,${EXTEN:${TRUNKMSD}}) Because of

How I can retrieve the original ${EXTEN} value when falling into the exten => i,1,whatever "context"?? I'm trying to implement this extension rule: exten => i,1,NoOp(${EXTEN}) exten => i,2,Wait(1) exten => i,3,Playback(vm-extension) exten => i,4,SayDigits(${EXTEN}) exten => i,5,Playback(is-invalid) exten => i,6,Playback(pls-try-again) exten => i,7,Goto(s,12)

I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help?

Hello all, I haven't used cisco phones as yet - do they work with asterisk, are they good which models are the best? I am after a starting point! Thanks Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/f5246238/attachment.htm

Hi All, I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: [codec_g729a.so] => (Annex A/B (floating point) G.729 Codec (optimized for i386)) Jul 19 14:11:23

Please can someone help me with the following two error messages: Error 1. I have loaded the Zaptel dirvers and everything is ok with ztcfg. I have configured Zapata.conf and everthing looks good but it apears the Zap channels dont load when starting Asterisk. When I make a call to one of the fxs port I get the following error message. -- Executing Dial("SIP/39-b204",