similar to: Help needed - ISDN is "redialling"

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

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 Marius, If you're really using Ubuntu 10.04 (lucid lynx), the r-base package does not exist anymore on the repository (http://packages.ubuntu.com/search?keywords=r-base). You're version of Ubuntu is obsolete. If you want an Ubuntu LTS, you should upgrade to 14.04. >> Message: 1 >> Date: Sat, 15 Aug 2015 16:19:33 +1000 >> From: Marius Barnsley <mariusbarnsley at

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

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

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:

Hello, Did you setup your sources as suggested here? http://cran.us.r-project.org/bin/linux/ Sent from my iPhone > On Aug 15, 2015, at 3:00 AM, r-sig-debian-request at r-project.org wrote: > > Send R-SIG-Debian mailing list submissions to > r-sig-debian at r-project.org > > To subscribe or unsubscribe via the World Wide Web, visit >

Asterisk SVN-trunk-r7353M (will be moving to 1.2.2 this weekend) E1 connected to Sangoma A102 SIP phones (Cisco 7960) I've been making a call from my mobile to the office, when, suddenly the conversation is terminated and replaced by a "fax-type" sound. This has happened to me several times over the past year, so it's not the version of asterisk (we've had cvs trunk and

I am using Ubuntu 10.04 and have tried to download R-base with the following command: sudo apt-get install r-base and the following happens: Reading package lists... Done Building dependency tree Reading state information... Done Package r-base is not available, but is referred to by another package. This may mean that the package is missing, has been obsoleted, or is only available from

Is there any way of running a diagnostic on a Sangoma A102 card ? Our lines have gone down and I want to avoid the usual BT "It must be your equipment" line with technical proof. At the moment I have a BLUE/RED alarm on zap show status Julian.

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

Nothing from me is posting to the list either. Julian

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

I have a new Sangoma A102 and I'm trying to get it running in asterisk. A look through the dmesg log shows the card is detected and the various channels created. However, when I start asterisk I get the error below. Any ideas? My zapata.conf is below. Thanks, MD == Registered custom function SIPCHANINFO == Registered custom function CHECKSIPDOMAIN == Manager registered action

Hello, I would like to convert an objects list such as objects() or ls() that outputs "a101" "a102" "a104" "a107" "a109" to read within a list statement as follows : list(a101,a102,a104,a107,a109) Thanks Tony

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

I have a problem that is probably so "doh" I will be embarrassed. However, I have spent all evening on this with no success: I have the following setup (asterisk cvshead as of today) 10 Channel EuroISDN<=>Asterisk<=>Meridian What I can do: Call from outside into the asterisk, dial an extension, and pass through to the meridian. WooHoo. What I can't do: Call from

In prepping our IMAP migration to Dovecot, I've hit upon a small issue: the "IMAP server directory" which Thunderbird users have set in the "advanced" prefs. Our old IMAP server seems to set your current directory to ~. This wound up causing users to set their IMAP server directory to "/home/<user>/mail/" so that their mailboxes would be found: --> A104

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