search for: sinergi

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2004 Sep 27
6
FXO question
Hi all: Does somebody know how many meters of cable is supported between Asterisk Digium-FXO cards and the analog telephone ? Thanks, Angel
2005 Jul 13
2
SMS over SIP and Asterisk ??
Hi, Is there a way to send and receive SMS over SIP protocol with Asterisk ? I mean, between two SIP phones like below... SIP_phone "A" (sending sms) ====> Asterisk ========>SIP_phone "B" (receiving sms) ... Is it possible ? If so, how could I do it ? Thanks, Angel. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 Sep 28
0
FW: FXO question
...on Asterisk Mailing Lists > [mailto:benjk.on.asterisk.ml@gmail.com] > Sent: Monday, September 27, 2004 11:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] FXO question > > On Mon, 27 Sep 2004 11:51:16 -0300, Angel Diaz <adiaz@sinergis.com> > wrote: > > Hi all: > > Does somebody know how many meters of cable is supported between > > Asterisk Digium-FXO cards and the analog telephone ? > > You don't plug analog phones into FXO cards, that's not what FXO cards > are for. > > r...
2004 Oct 01
0
Re: [Asterisk-Dev] Use the Meetme application with another module thanUSB-UHCI
...`insmod zaprtc` because zaprtc is a replacement for rtc. I've had excellent success in recompiling our 2.4.8-20SMP kernel (standard RH9 kernel) to have RTC as a module and now I have full MeetMe abilities on our PE. Matthew ----- Original Message ----- From: "Angel Diaz" <adiaz@sinergis.com> To: <asterisk-dev@lists.digium.com> Sent: Thursday, September 30, 2004 12:32 PM Subject: [Asterisk-Dev] Use the Meetme application with another module thanUSB-UHCI Does somebody know why the ztdummy use the USB-UHCI ? What functionality of the USB is used ? Is it possible to use wi...
2002 Oct 15
1
Réf . : Re: trying to use wondershaper on a dedicated line (not adsl)
here''s the output : # tc -s -d class show dev eth0 class cbq 1: root rate 10Mbit cell 8b (bounded,isolated) prio no-transmit/8 weight 10Mbit allot 1514b level 2 ewma 5 avpkt 1000b maxidle 23us Sent 294 bytes 7 pkts (dropped 0, overlimits 0) borrowed 0 overactions 0 avgidle 605 undertime 0 class cbq 1:10 parent 1:1 leaf 10: rate 100Kbit cell 8b prio 1/1 weight 100Kbit allot 1600b
2005 Feb 04
2
Swap Memory get used totally
Hi list, Time to time, my asterisk goes down.Verifying with TOP, I see the swap memory of the computer get used totally but, I don't see what the process is using it. Hereis a copy wath I see doing top. Does somebody have an idea ? My asterisk version is ====>>> Asterisk CVS-HEAD-08/18/04-22:30:24 Thanks Angel. 08:49:19 up 5:23, 1 user, load average: 0.50, 0.70, 0.64 35
2004 Sep 17
2
Re: Asterisk-Users Digest, Vol 2, Issue 163
Hi Matt, I have verified with ztmonitor the audio level and it was too low, then with this the fax machine report "Not Response". I modified the audio level in zapata.conf and after that the fax machine report "Commnunication Error". Do you an idea what could be ? Thanks, Angel. > Message: 3 > Date: Sat, 18 Sep 2004 00:48:23 +1200 > From:
2005 Jul 06
11
Connect 30 phone lines to asterisk how to
Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel.
2005 Feb 22
0
VMS - AGI
Hi list, I would like to use the * VMS application with a GSM network, I know that * support Unavailable and Busy Redirecting Reason in the extensions.conf but, what's about the No Reply ?? Then, I know that doing a debug on the PRI, I see the redirecting reason... My question is; is there anyway to obtain this redirecting reason through perl AGI ?? Thanks in advance . Angel.
2005 Mar 15
0
Voicemail Question...help
Hi all, Can anybody help me on this ? I'm trying to use asterisk voicemail application. The thing is, when a call forward from the switch arrive to asterisk over ISDN PRI I verify if the redirecting number match one subscriber of the asterisk voicemail databases otherwise, forward this call to another voicemail platform. My question is, how I have to forward this call to the other
2005 Jul 11
1
VoIP services
Hi guys, Can somebody help me on some questions please ? If I have a VoIP network with my Asterisk platform in Europe, what do I need to interconnect my VoIP network to another network in the USA in order to my customers in Europe be able to call to customers in the USA network ? The network in the USA is not an Asterisk platform. Thanks,
2004 Nov 16
10
SS7 for *
Hi all, Does somebody know what's new with SS7 and * ? I'm very interested. Is it ready ? I'm prepared to pay if necessary. Thanks, Angel.