search for: verastegui

...oss and a very high jitter. I tried several codecs and parameters, and the only thing left to test is PLC (Packet Loss Cancellement). Have the astesrisk and digium people implemented PLC?, Are they implmementing it now? and, if not, Where can i find an implementation? Thanks in advance -- Jorge Verastegui G <jorge@redcetus.com> RedCetus S.R.L.

...or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. Please tell me how to solve this issue Thanks in advance Jorge -- Jorge Verastegui <jorge@redcetus.com> RedCetus S.R.L. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040516/fbd6bc30/disclaimer.html

....gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040419/665c0fc6/disclaimer.html

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

...f codecs were proven ( ulow, g729 ). When use the Asterisk with Sip phones everything works well. SipPhone------>Asterisk------->PSTN(B) The configurations, are the usual ones (from the wiki). the version of asterisk is 1.0.3, the linux is FC2. Please help me. -- Jorge Verastegui G <jorge@redcetus.com> RedCetus S.R.L.

...created the file: /var/lib/va-infoclient which contains your machine signature and that you must send to Voiceage to obtain a valid certificate for the g729 library. Of course my Internet connection is OK, (no proxy) How can i solve this problem? Thanks in advance Jorge -- Jorge Verastegui <jorge@redcetus.com> RedCetus S.R.L. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040515/e9ec9698/disclaimer.html

...ll work with this PCI Express. <http://www.digium.com/index.php?menu=wildcard_te405p> Has anybody worked with PCI-e yet? As far as i understand, the Wildcard TE410P <http://www.digium.com/index.php?menu=wildcard_te410p> (3.3 volts) will not work. am i right? thanks in advance Jorge Verastegui RedCetus.com

...es echocancel=yes echocancelwhenbridged=yes rxgain=1.0 txgain=1.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=4 group = 2 switchtype = national signalling = pri_cpe channel => 1-15 channel => 17-31 channel => 32-46 channel => 48-62 Jorge -- Jorge Verastegui <jorge@redcetus.com> RedCetus

...= 0 musiconhold = Default busydetect = yes busycount = 4 jitterbuffers = 8 relaxdtmf = yes callwaiting = yes usecallingpres = no callprogress = no threewaycalling = yes transfer = yes cancallforward = yes callreturn = yes group = 3 callgroup = 1 pickupgroup = 1 channel => 49-52 -- Jorge Verastegui <jorge@redcetus.com> RedCetus

...43151 IO-APIC-level uhci_hcd, libata, eth0 19: 510477191 IO-APIC-level uhci_hcd, wctdm 22: 510396103 IO-APIC-level t1xxp NMI: 0 LOC: 510532759 ERR: 0 MIS: 0 I am thinking to replace the two T1 cards by a new TE205P Thanks in advance for any comment Jorge Verastegui redcetus.com -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-keys Size: 1328 bytes Desc: PGP Public Key Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/bede5b8e/attachment.key

chan_sip.c6524 reload_config= unable to get ip address from asterisk, sip disabled The ip address is working fine, Internet works great. Can anyone help...Thanks

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

We would like to know if you have any recommendations for softswitchs to be used by a small size telco with sip services in *.

Hi I have an asterisk box with digium hardware connected to a Siemens EWSD version 15 using a crossed E1 cable. The asterisk is serving as a h323 media gateway. When i start asterisk, the Siemens gives me this alarm: REPORTES GENERADOS EN LA EWSD AES/V15SBOL/BOLCBK1V51327079/013 05-06-06 11:25:38 7773 3062/03728 HF.ARCHIVE-80040