similar to: Conneting Asterisk to Swyx pri

Hi, Does anybody have an idea as to why dahdi_test results drop to unacceptable levels after doing a wanrouter stop/start using a Sangoma card? See below that it drops from 99.99% to 98.55%: [root at bin]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.999512% 99.992874% --- Results after 2 passes --- Best: 100.000 -- Worst: 99.993 -- Average: 99.996193, Difference:

Hi, I'm a student and my thesis work consist in testing Asterisk with Swyx(SwyxWare). My approach is to declare asterisk as h323 gateway for the Swyxserver using oh323 Plugin. Is there any possibility to connect Asterisk with Swyx? how? the outgoing call must pass from Swyxit->to Swyxserver-> Asterisk->to PTSN Thanks

Hi all, My problem boils down to these errors: ... Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time This is triggered by lines in extentions.conf such as: exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W) The system is CentOS v5.2 with Asterisk 1.4.23 (druid-asterisk-1.4.23.1-2), a Sangoma A104

Hi, I'm using Sangoma's wanpipe together with dahdi, all software downloaded today at most recent version. Hardware is Sangoma A104, a 4xE1 card. Installation went well. Anyway, wanrouter status shows a different result than dahdi_tool or dahdi_scan. I've just put a hardware loop on port 1. All the other ports are open. wanrouter status shows the expected result: Device name |

Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create

Greetings folks- I'm experiencing issues with a freshly installed box. When a call comes in via PRI (Sangoma AFT-A104), I see this in my logs: [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 12 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (Invalid argument) [Dec 15 14:26:10]

I'm trying to send CallerID info to a MetaSwitch system over a PRI. The MetaSwitch gets the info exactly as it is sent by Asterisk, but I think it might be having trouble with the Hexadecimal b1 that is being sent just before the first character of the CallerID Name. Does anyone know what the significance is of the b1 being sent here? Or, is there a way to make Asterisk not send the b1

I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0 on FC7 all went okay. using sangoma a104dx on both machine. I followed the write up on http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+setup I have the cross over cable between them. however, wanpipe shows connected but the signaling link does not align. i have my configs for host A ##wanpipe1.conf [devices] wanpipe1 =

Hello, I have following problem with Sangoma A104 card: CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 NOTICE[16509]: chan_zap.c:7494 pri_dchannel: PRI

I need your help I have a sangoma A104D on my dell server; I got card status ok with no alarm If I dialed the extension 6210006, it shows the output as stated below, but there is no ringing from the pstn number nor the iax softphone am using on my pc. I will be glad if someone can give me a working config? What I want to achieve is to send all my call to the pstn on A104D? The pstn am

Hi there, we have a problem with dahdi and overlapdial. We are running an E1 in Germany and are in need of overlapdial. The E1 is connected to a Sangoma A101. As soon as overlapdial is set to "yes" we have problems with incoming audio on the dahdi channels. When set to "no" all audio is fine. Basically we can choose between being able to receive calls or to place calls

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

I upgraded from Tormenta2 to Sangoma A101. I followed the instructions, and installation was successful. zttool, ztcfg, all show card is installed properly. I copied the parameters from my old working zaptel.conf, zapata.conf and zapata-auto.conf. Verified on Sangoma website that these files are correct. Also configured wanpipe1.conf. But doing all this didn't start the PRI channels. It says

Hi, I am trying to validate a setup with a sangoma A116 card (16PRI in one card). I currently have two machines set up, each with a sangoma A116 card. Those are interconnected with crossed PRI cables. One of them is in NT mode, the other in TE. The ports are configured in E1 mode, asterisk is all set up and I can make calls between them, filling up all 480 (16 * 30) channels. Now, I learned

Hey Guys! Please help me to find out issue. I have two PRI ## Span 1: WPT1/0 "wanpipe1 card 0" span=1,1,0,esf,b8zs bchan=1-23 hardhdlc=24 echocanceller=mg2,1-23 ## Span 2: WPT1/1 "wanpipe2 card 1" span=2,2,0,esf,b8zs bchan=25-47 hardhdlc=48 echocanceller=mg2,25-47 Sometime my calls got through but some time i am getting pri cause 44 sebpbx1*CLI> == Using SIP RTP

System have been working great for weeks, using an average 40 of 120 dahdi channels. But today, I suddenly see scary things like this: -- Moving call from channel 5 to channel 7 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing

Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten => 12345678,1,Answer() exten => 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -----Original Message----- From: Erick Perez [mailto:eaperezh at gmail.com] Sent: Thursday, July 26, 2007 7:03 AM To:

Hi, in a PRI setup, the receiving side is changing the B channel at proceeding. It seems this sometimes breaks some logic (pri_fixup_principle) and then the hangup kind of breaks, release is not answered and a restart cycle is triggered (by remote side). Anyone can help me debug this ? I've seen many posts with simmilar issues but no answer/solution. This is happening on Asterisk 1.2.16 +

Hi, I have a client that has a 24 channel voice T1 that I have been using e&m signalling over for a number of years. The local telco finally got an ISDN switch and wants to move them to PRI. I didn't see this as a big problem - I've done a few others on this particular Caribbean island without issues, but this would be the first time with a Xorcom unit involved. We tried to do

Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection ....now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a