similar to: Connecting a Rolm CBX to Asterisk via T1?

Okay, T1 card issue sorted out. New Lesson: Stay Away from TigerJet chips. Next up, modem pool -- I wanted to know if the below config looked anywhere near half-sane for defining in asterisk what is essentially a small pool of four waiting modems that will handle faxes if another modem is busy: exten => _X.,1,Dial(IAX2/iaxmodem0/${EXTEN}) exten => _X.,2,Busy exten => _X.,3,Hangup

Hey all, Working slowly on getting the myriad number of parts to my fax system plan together, and one of the pieces I want to nail is how to go about, for the outbound context (fax-out) pre-pending a digit onto a number? I.e., for all my testing right now, I've been dialing '91XXXXXXXXXX', as the asterisk server doing faxing junctions into my old Rolm CBX switch, and so I need the

Hmm, I don't know if the zaptel fax detection will trigger. This is going through my company's Rolm CBX switch, using a plain T1 cable (yanno, RBS, E&M Wink, and all that jazz) between the two systems as a Tie line, so that may mess with the fax stuff. The Rolm's just wired to take a special extension block and pump anything coming in on them out the T1 trunk group to the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, 2 weks ago I asked questions about PJSIP and T.38 but got no replies. I upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having the same issues. In the trace below, I'm sending a fax from Hylafax server through iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw) connected to the PSTN via ISDN; the

I am currently a new asterisk user I have worked with the old rolm systems in the past. I have been asked to look around and find out how to do a few things in asterisk, either in asterisk itself or with third party software. The features that I am looking for are: 1. A good management application for setup and edit the .conf files for both the exten & voicemail. 2. A receptionist software

There's a #define macro in channels/chan_iax.c that you can modify to make this forced value higher. Just open it up in your favourite editor and search for '60' and you'll find it. Now if there's an easier way than having to change a source-level macro, I'm all ears... Cheers!, --jkinard -----Original Message----- From: asterisk-users-bounces at lists.digium.com

I have setup hylafax today, along with iaxmodem. I'm just starting the debugging process and see the following message every 60 seconds: [Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry: Restricting registration for peer 'iaxmodem0' to 60 seconds (requested 300) Can someone tell me what this means? Why is it there? And how do I get rid of it! Thanks, MD

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

hi folks. i'm experimenting with iaxmodem + hylafax using DID to determine where to send the fax to it's final destination. however i have difficulties passing the DID information from iaxmodem to hylafax. in extensions.conf: exten => _XXXX,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r) exten => _XXXX,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r) exten => _XXXX,n,Busy exten => _XXXX,n,Hangup

What an ACTIVE newsgroup! I'm in the early stages of researching Asterisk. My current environment is a small college (~1000 sets/~400 student sets), Avaya Definity G3si/Seimens Rolm Phonemail. As you can imagine, the maintenance, licensing, and equipment costs are HEFTY. So.. are there any small colleges/universities using PBX or Voicemail? If so, I'd be interested in your migration

Hi all, Noticed a curious issue in my testing setup for a faxing system I'm putting together, but it looks like if I let the lines all sit idle for a few days (no one uses this yet, so the whole thing really does sit idle until I do testing on it or something else), something I believe on my Asterisk end goes to a kind of "Sleep". It's hard to describe really, but I'm not

Hi there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a ?crit : > I think the "488 Not acceptable here" is occurring because the channel > connecting through is not T.38 capable, that will be the IAX channel > from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more

I am having a problem with and Asterisk installation where two ports connected to a TDM400 card will have a very loud noise when you try to dial. The server has an OpenVox D110P, a TDM04B and a Xorcom Astribank 8 fxs. It is running Zaptel 1.4.11 and Asterisk 1.4.18. The problem always happens with two ports (34 and 35) which are connected to two GSM gateways. They will work fine for a week

Hi everybody. I have an HP StorageWorks array connected to one of my servers, connected via an LSI controller (lspci reports: SCSI storage controller: LSI Logic / Symbios Logic SAS1068E PCI-Express Fusion-MPT SAS (rev 08)) on an HP DL385 running CentOS 5.5 x86_64. I have set up multipath and all is working but, when I create a snaphost and export it as a new LUN from the HP web interface, it

I'm having problems with PXELINUX passing through to a "localboot" on a HP Proliant DL385. I'm working with 3.73. If I configure PXELINUX to directly "localboot", everything works. If I configure PXELINUX to use menu.c32 or anything like that, a subsequent localboot fails. I see the expected "PXE ROM exiting" message but the next device doesn't

Hi there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P

I am getting a lot of winbind errors in my logs on one server. Any idea's? Apr 7 10:32:19 pdc winbindd[8789]: [2011/04/07 10:32:19.062866, 0] winbindd/idmap.c:201(smb_register_idmap_alloc) Apr 7 10:32:19 pdc winbindd[8789]: idmap_alloc module ldap already registered! Apr 7 10:32:19 pdc winbindd[8789]: [2011/04/07 10:32:19.063011, 0] winbindd/idmap.c:201(smb_register_idmap_alloc) Apr 7