similar to: TDM400P FXO lines problem

I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has

Currently running two POTS lines into an asterisk system. Analog and SIP on premises. Being in the sticks, the POTS service is abysmal for quality, especially in the rain. Recently, cable has become available with VOIP phone. The cost savings are attractive as it can replace several independent services for TV and internet (currently satellite). But, I cannot get much out of them, regarding

Please chime in if you've ever wished for digium to make a 4-port daughter board with a combination of 2FXO AND 2FXS ports on the same card. When using the 800 series cards, one must either choose 4-port permutations of FXS/FXO, OR one must give up 2 valuable ports. In other words, when you add ONE 100-series daughter board, you give up TWO of your physical ports. Is there a technical

I have set up asterisk@home <mailto:asterisk@home> with Digium TDM400P 2FXO/2FXS. I am unable to seize my trunks from either soft or analog phones. Inbound calls result in answer/disconnection. I see the following error code on my asterisk server INIT: Id "s0" respawning too fast: disable for 5 minutes Does anyone have any suggestions for me? I?d really appreciate some help on

I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). Before I implemented Asterisk, some users were using Bell services to set-up 5 way calling: The user would set up a three way call on one line, switch to the second line, set up another 3 way call and then link the two lines together with the Flash key, thus establishing a 5 way call (the user, 2 others on line 1, another 2 on line 2). How

I'm trying to replace procmail with deliver but I'm having problems with the group part of things. I keep getting: Aug 29 16:44:19 dingbat deliver(mimo): open(/var/mail/.temp.dingbat.3969.d1689935a308e0dd) failed: Permission denied Aug 29 16:44:19 dingbat deliver(mimo): file_lock_dotlock() failed with mbox file /var/mail/mimo: Permission denied My primary group is mimo and I think

Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing

Hello, I connected a Micronet SP5014 2FXS + 2FXO gateway to the asterisk, the problem is i can make call but can't receive calls. If i make a "sip show peers" it shows the micronet is not connected to the asterisk. Does anybody knows how to configure the micronet and asterisk to solve this problem ? Thank you

I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power will this drain from the 12 and 5 V connector when all ports are in use? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ ------------ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . -------------

Hi I've a Sangoma A200D with 2FXO and 2FXS. When using it with only the FXO module, it's all good. But when I put in the FXS module and connect the power, logs tells me not enough power. > Mar 31 14:11:54 phone kernel: [ 4761.246931] wanpipe1: Module 1: > Failed to powerup within 600 ms (8V : 72V)! > Mar 31 14:11:54 phone kernel: [ 4761.246937] wanpipe1: Module 1: Did

Hi Evan > > It looks > > like that the instruction selector operates on actual DAGs, no > > unDAGing to > > trees seems to occur at any point. > Instruction scheduler is responsible for turning a DAG into a list of > instructions. So unDAGing is applied by the instruction scheduler. At which points in the compilation flow is the instruction scheduler run (i'm

With the lack of info on Yoda Communications in Taiwan and their hardware, I thought I'd post my experience. I got my hands on a few H.323 VG-400's and VG-100TA's. http://www.yoda.com.tw/model.php?type=VoIP_Solution&pname=VG400 2 of the VG-400's were 2FXO/2FXS models. A couple were deployed to SE Asia, where we planned to offer our services. Originally, I ran a GnuGK server

Hi, I'm trying to get the drac plugin working (again) but I'm stuck with the following message: Restarting IMAP/POP3 mail server: dovecotEDRAC: net_ip2addr() failed: Operation not permitted This is on a Debian Squeeze box, dovecot version 1.2.15-7, drac version 1.12 Looking at the source it should print the IP in the brackets of net_ip2addr. The fact that it's just empty made me

I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN

Hi all, A while back, there was a short thread on using the FXS interface from a Primus TalkBroadBand device (a DLink ATA) as a incoming line for the FXO interface on the TDM400P: Primus <--> DLink ATA FXS <--> TDM400P FXO <--> Asterisk In that thread, a couple of people suggested that this does not work reliabley, and the ATA FXS <--> TDM FXO link 'goes

Sorry in advance if this is slightly off topic. I'm selling a Digium TDM400P with 4 FXO ports for $300 plus $20 shipping costs. This is an authentic Digium TDM400P. It's brand-new and was only installed into a system once and then removed (we decided to go with T1 spans instead). Please email me off list if you're interested in purchasing it. Thanks! -- Kenneth Shaw ExpiTrans, Inc.

Not to sound like a broken record but...... Is there any word on the availability of the FXO cards for the TDM400P? I have an application that would benefit. If it has been dropped please let me know. Thanks Gene Kochanowsky

Since the irc channel wasn't any help, I will try posting my problem here. I have two TDM400Ps less than a week old in a PC. All of the FXS ports work fine, and all of the FXO ports worked fine up until thisafternoon. If I try to dial in, I get a busy signal, if I try to dial out, all I hear is a very scratchy, very crackly dialtone. If I swap the first FXO module with the second on the card,

Hi Guys, I recently got a TDM400P 4 FXO for use in the UK, this at the time seemed like a good idea as I had good results with an X100P clone. Installation went great and call clarity is excellent no echo like I had on the clone card. My problems start with detecting hanging up the line. If a person calls into the system and speaks to me on a SIP phone when I hang up the call clears

I've setup * with TDM400P w/1 FXS, 3 FXO modules. I've one analog phone connected to TDM400P FXS module, 1 PSTN line to one of the FXO module(ZAP) , and 2 analog phones connected to Sipura 2000 (SIP). The calls between SIPs and zap phone (fxs) are OK. But 2 issues cannot be solved: 1. To dial to PSTN via zap phone, the setup in extensions.conf with the following exten =>