similar to: A102 card, BT ISDN30e, silence

We are trying to configure a Sangoma A101 card to allow both incoming and outgoing calls on a UK (BT) ISDN30e line with only 24 channels enabled. At present incoming calls work fine. We can't call out -- we get a BUSY/CONGESTED error. Do we need another context in our zapata.conf? In other words, do we need to reserve, say, channels 17-24 for outgoing calls? I also wonder if the signalling

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

Hi All, Has anyone hooked up * as an extension/trunk of an Avaya system that has around 2 ISDN30e's. Trying to add 100 extensions to one of our systems, but not sure where to start reading. Thanks. -- Kind Regards, Gavin Henry.

Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only

Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31

Hello, Have you tried the wanpipe-beta8c-2.3.3.tgz release in the custom/2.3.3 dir on their FTP site? Also, have you contacted Sangoma for support? They are very responsive. I am using wanpipe-beta8a-2.3.3.tgz and it's been working great on my A104 for a week now. MATT--- -----Original Message----- From: Dmitry Zhukovski [mailto:DZH@comx.dk] Sent: Monday, May 09, 2005 5:20 AM To: Asterisk

Hi again, Well - I didn't see beta8a-2.3.3 in custom dir. Will try. Also I tried to contact Sangoma - they are very fast to answer but main problem is time difference - it's 6 hours between Canada and Europe. Br, dmitry Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-2600 Glostrup Denmark Phone: +45 70 25 74 74 Fax:???? +45 70 25 73 74 Web: www.comx.dk

Ok, I have tested with almost all versions both in 2.3.2--*-stable and 2.3.3-*-beta - I am getting same messages: May 9 10:55:26 WARNING[3961]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! and same Down state pb01*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7

Shortly, I'll be purchasing a Sangoma A102. I'm wondering if I should spring for the hardware echo cancellation circuit or not. Upon initial implementation, the 2 T1 Ports will be used as a passthrough as we slowly transition off of a legacy PBX. Eventually, we'll only be using one of the ports, and will be providing VoIP service to a bunch of SIP deskphones. So - with that usage

Hello, in our company we are trying to do this: Fax <--> Traditional PBX <--> Asterisk <--> PSTN In practice, we have put an Asterisk equipped with a Sangoma A102 (2 PRI ports) between our PBX (Siemens HiCom) and the PSTN in order to have a VoIP network along the traditional telephony network. The problem is with the fax. We just want to send and receive faxes from/to our fax

I think you are absolutely right. The audio I heard earlier sounds exactly like a timing issue. So: wanpipe1.conf: TE_CLOCK = NORMAL TE_REF_CLOCK = 0 wanpipe2.conf: TE_CLOCK = MASTER TE_REF_CLOCK = 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs I'm going to make this change and reload at lunchtime, I'll document it and post it to the list if it works.

Followup on this issue, it appears that using a single PRI's clock as the master clock avoids clock drift between the PRI's and we get no more artifacts. So, : wanpipe1.conf: TE_CLOCK = NORMAL TE_REF_CLOCK = 0 wanpipe2.conf: TE_CLOCK = MASTER TE_REF_CLOCK = 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs -----Original Message----- From: Michael L. Young

Any benefit on getting the PCI Express version? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070222/62f79910/attachment.htm

Hi! I've went through the READMEs and could not answer this question: During installation, the Setup program asks: Would you like update/upgrade wanpipe drivers? (y/n) For a pure Asterisk TDM installation - is it required to patch the kernel or is this only when using the sangoma cards as WAN router? regards klaus

I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: "very bad phasing reverb & feedback (from my rock & roll days)". This is quite intermittent, as in most cases, the user

Aha, it just happened to me, so now I can characterize the audio: It basically sounds like it's missing every other sample - fuzzy and distorted. Timing?

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:

Does anyone have any experience ith configureing the sangoma A102 card for testing using a e1 cross cable i've configured and installed the cards properly even the lights on the card are green which proves that my cross cable is properly built too. my problem is with asterisk which gives me these errors PRI got event: HDLC Abort (6)on Primary D-channel of span 1 PRI got event: HDLC Bad FCS

We will be connecting our Asterisk server to ISDN 30 and intend using the Sangoma A101 card. The install location is in London (UK). Sangoma card at Voipon http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA I would be grateful to hear if this is the right choice of card. Usage reports would be helpful. Regards Rory -- Rory Campbell-Lange

Hello, I'm trying to connect a Digium Wildcard TE110P T1/E1 Card to a 10 channel BT ISDN 30 box, however I'm fairly new to Asterisk and this is the first time I've set-up a telephony interface card can anyone give me some pointers or known working configurations? I'm running Asterisk 1.6 and Dahdi 2.4.1.1. Thanks, Tanuki -------------- next part -------------- An HTML attachment