similar to: Re: Sangoma A102 cards testing FIXED

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

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

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:

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

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

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

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

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

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

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?

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

I just installed Festival 1.9.5 and all dependencies. It works properly from the command line, as I can play the contents of a text string or file from the Festival command line through PC speakers. I then added the code to the festival.scm file, as detailed in Method 1 of http://www.voip-info.org/wiki-Asterisk+Festival+installation I did not apply the patch used in Method 2, because I can not

Hello, Several people asked to get a hold of the stats I used to determine that the Sangoma T1/E1 boards performed better in our real-world tests than the Digium boards. We've decided to post our results after confirming them over the past month of operations. Here is a link to the last two reviews I wrote about Sangoma with Asterisk:

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

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

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

Anyone using a Sangoma A102 with a Dell 750? We are looking at going this route but needed some input. I really only need a Single T1 port, but this server doesn't have a PCI-X port, which the A101 apparently requires? Thoughts, Suggestions? K

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.