Re: [asterisk-users] Zaptel to Dahdi
Converting is actually pretty straightforward: Bare minimum: /etc/zaptel.conf -- /etc/dahdi/system.conf /etc/asterisk/zapata.conf -- /etc/asterisk/chan_dahdi.conf Any reference to ZAP/* becomes DAHDI/* in your asterisk conf files (i.e., extensions.conf). Granted, all I use Asterisk for is a fax-to-email mechanism in conjunction with my ~18yr-old Rolm system, but I imagine more complex setups are probably not too hard to replace. Most of it was just search replace in the extensions.conf file. You can also leave the older zapata.conf intact. I believe newer Asterisk version will ignore the file's existance. Ditto on the older zaptel.conf, since the dadhi code doesn't even reference it I believe. The only thing I miss, is I thought Zaptel was a pretty cool name. Like, lasers shooting everytime a call comes in or something. Dahdi makes me think of angels singing everytime a call comes in now. HTH, --J From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Sunday, April 19, 2009 11:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Zaptel to Dahdi VoIP-wiki.org states : Digium resources http://www.asterisk.org/zaptel-to-dahdihttp://www.voip-info.org/wiki/edit.php?page=http%3A%2F%2Fwww.asterisk.org%2Fzaptel-to-dahdi /etc/zaptel.conf Becomes /etc/dahdi/system.confhttp://www.voip-info.org/wiki/view/system.conf /etc/asterisk/zapata.conf Becomes /etc/asterisk/chan_dahdi.confhttp://www.voip-info.org/wiki/view/chan_dahdi.conf Now, what do I have installed on my system : /etc/zaptel.conf and /etc/asterisk/chan_dahdi.conf Will these two config-files work together ??? I have no /etc/asterisk/zapata.conf and no /etc/dahdi/system.conf Do I create an empty zapata.conf ?? I also do not have /usr/lib/asterisk/modules/chan_zap.so !! My Asterisk-version : 1.4.24 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi Init script for Suse?
Doesn't look like SuSE is that evolved just yet. I poked at a few other init scripts in /etc/init.d, and they're all pretty much in the format of: echo -n Starting something ... command rc_status -v Some of the init scripts are downright horrific in their design because of this. I would imagine the newer SLES stuff might have cleaned things up, but Open Enterprise Server is based on SuSE 10.1/10.2. I would imagine that, whenever Novell gets around to it, OES3 will probably be based on SuSE 11 (if they even have that version -- I haven't checked yet). I think they're still ironing out kinks and trying to force stubborn holdouts (like myself) off of NetWare onto OES2. --J -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Monday, January 26, 2009 4:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi Init script for Suse? On Sat, Jan 24, 2009 at 04:08:53PM -0500, Joshua Kinard wrote: Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box that'll work right? The one included by default only deals with debian and redhat, and the changes between the old zaptel script I have that works are far too invasive. Notably in the use of this action command that's probably redhat specific. Anything equivalent in SuSE? The Debian equivalent of action Starting FooBar foobar , if you source /etc/lsb/init-functions is log_daemon_msg Starting FooBar foobar log_end_msg $? Though it has an atvantage of making it easier to redirect output of the command. (Those functions don't seem to be part of the standard LSB init functions, sadly) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi Init script for Suse?
Nah, not using RPMs. This is a from-source build. Part of the problem is, I'm running Novell's Open Enterprise Server, which is SLES 10.1 (I think) with Novell's OES2 Overlay on top. So the Asterisk RPMs available for that revision of SLES weren't usable for me (they had 1.2 available, and I wanted 1.4), hence, source build. Might play with 1.6 when I start feeling adventurous again, as my asterisk config is really, really simple (it's a bridge between an old Rolm CBX and HylaFax). But I figured, if their init script supports two of the top distros, why not a third? Wasn't sure, so I was seeking comments on the script I attached to my last e-mail. --J -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Saturday, January 24, 2009 7:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi Init script for Suse? On Sat, 2009-01-24 at 23:45 +0100, Marco wrote: Hi, I've it up and running on OpenSuse 11. I used the scripts provided by the sources and commented out one line: # # Determine which kind of configuration we're using # #system=redhat # assume redhat system=debian # assume debian This forces the script to use debian style. It works for me, except, if I remember well, some little problem on reload (but stopping and starting again works fine). Best regards, Marco Signorini. Just wondering... The O.P. said he's using SLE. You're talking about openSUSE. Are you using the rpm's from the OBS? The zaptel-rpm for 10.3 were containing the proper startup scripts. I've got some suse machines running asterisk, but as soon as hw get's involved, i'm stuck: neither pri, nor bri (mISDN) seems to be working on anything later than 10.3. Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi Init script for Suse?
Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box that'll work right? The one included by default only deals with debian and redhat, and the changes between the old zaptel script I have that works are far too invasive. Notably in the use of this action command that's probably redhat specific. There's practically zilch on google on the matter. I think suse support should be included by default, though. Thanks!, Joshua Kinard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi Init script for Suse?
Stared at an init script long enough, and managed to devise up the following script. This applies straight to tools/dahdi.init in dadhi-linux-complete. Minus the top hunk in the patch (which sets system = suse), this converts it into a working script for suse systems. Thoughts? What's the likelyhood something like this could get included in an actual release? If possible, what extra work needs doing? --J From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Marco [marcota...@libero.it] Sent: Saturday, January 24, 2009 5:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi Init script for Suse? Hi, I've it up and running on OpenSuse 11. I used the scripts provided by the sources and commented out one line: # # Determine which kind of configuration we're using # #system=redhat # assume redhat system=debian # assume debian This forces the script to use debian style. It works for me, except, if I remember well, some little problem on reload (but stopping and starting again works fine). Best regards, Marco Signorini. == INGEGNI Tech S.r.l. http://www.ingegnitech.com Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box that'll work right? The one included by default only deals with debian and redhat, and the changes between the old zaptel script I have that works are far too invasive. Notably in the use of this action command that's probably redhat specific. There's practically zilch on google on the matter. I think suse support should be included by default, though. Thanks!, Joshua Kinard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users dahdi.init-suse.patch Description: dahdi.init-suse.patch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Seconded/thirded too. Went from 1.4.18 to 1.4.19, stopped using -c and went to background and connecting using -r, and colors disappeared for me as well. I'm using screen as well (ls -l --color=auto works fine in screen too). Is there a documented fix available, or is this more just an odd curiosity regarding termtypes? --J -Original Message- Sent: Wednesday, April 09, 2008 10:16 PM Tzafrir Cohen wrote: On Wed, Apr 09, 2008 at 08:00:38PM -0400, Mike wrote: Ah, not bad. When I start asterisk with /usr/sbin/asterisk -c I get the colors, but if I start it without -c and then connect to the console using /usr/sbin/asterisk -r I get no color. Since I want this to be running in the background, how do I fix this so I get to have my cake and eat it too? The patch is rather trivial. Just make Asterisk pretend that it is vt100 (or whatever) if it is running as a service. I cant get color using asterisk -r on 1.2.17 or 18 either. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Hmm, interesting, initially it wasn't working. Maybe I started it from outside of screen? Odd. BTW, is it possible for the SuSE script to support a variable to pass args to the daemon? Like perhaps modifying ASTARGS to be a changable param at the top of the script? I like the verbose output, and attempting to add it there myself (and change Line 71 to recognize its existence) didn't pan out right. Right now, I have to manually send core set verbose 999 when connecting in. Thanks!, --J -Original Message- Sent: Thursday, April 10, 2008 12:10 PM Joshua Kinard wrote: Seconded/thirded too. Went from 1.4.18 to 1.4.19, stopped using -c and went to background and connecting using -r, and colors disappeared for me as well. I'm using screen as well (ls -l --color=auto works fine in screen too). The colors work if you use the supplied init scripts. cd /path/to/asterisk/source/contrib/init.d Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors
-Original Message- Sent: Thursday, April 10, 2008 1:08 PM Or maybe it's plain buggy. Bug reports are welcomed. Nah, I think it was PEBKAC and PICNIC here :: sheepish grin :: I tried adding --style options to the DAEMON var assignment, and it looks like the -f check further down didn't like that. I'll look into the sysconfig setting...more used to Gentoo setups than SuSE. BTW: why do you prefer it to start verbosely? This tends to clutter the logs with useless information. Oh, this is just for a faxing system. We have an ancient Rolm in place for the actual phone calls. I want to monitor/log things when I roll it out for testing to catch any oddities that may occur with inbound and outbound faxes. Communication between it and the Rolm sometimes went to sleep (I sent a mail here on that, but got no responses), and so, I want to watch for it in case it happens again. Might've been a bug fixed in the newer releases of zaptel and asterisk (which I'm running now). --J ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors
-Original Message- Sent: Thursday, April 10, 2008 1:22 PM 15? What do you need that for? IIRC the highest verbosity level is 5. anything more than that doesn't change the clogging of your logs. Ah, 5 is max? Kinda like gcc not supporting anything greater than -O3? Good to know that. I figured - was overkill, but I hadn't dived into the options parsing code to actually verify that. --J ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Config file for 'make menuselect' available?
Hi all, Was curious to if for the 'make menuselect' command, there's a config file hiding someplace that lets me quickly move a configuration to a new source tree (much like .config in the kernel trees). I looked around after running menuselect and compiling, but none of the files stood out as config files, so I thought I'd put the question to the list. Thanks!, --Josh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk (or maybe Zaptel) goes to sleep after inactivity?
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 sure if it's Asterisk itself or maybe the Zaptel side of things. The whole system is connected by a T1 Tie line to my old Rolm system (a Plain T1...RBS, EM Wink, etc..), and on the Rolm side, when this sleep issue crops up, my monitoring there either shows things like RING-IN or ERROR, but no connection is actually made between the Rolm and Zaptel, so it's like the Rolm is getting a signal down the line, but not a signal it likes. I see the Wink indicators, so I doubt it's the signalling or anything. On the Asterisk side, it just says Everyone is busy/congested at this time (1:0/0/1) almost immediately after Dialing out, and then plays the no-service message. But once you make a couple of call attempts on the line after a few days of inactivity, either fax calls via HylaFax or via an IAX softphone I setup for testing, it all just starts working again, which really baffles me. Lately, I also found shutting down Asterisk and restarting zaptel seems to work too. It's kinda like getting out of bed after taking a long napalmost like the system is groggy and so poking it with a stick (repeated call attempts) or hit it with a bucket of water (restarting Asterisk/Zaptel) wake it up (unfortunately, my systems don't drink coffee). Thoughts perchance? Asterisk version is 1.4.18.1, and Zaptel is 1.4.9.2. Thanks!, --Josh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
I've got two DL385s and a DL320, and they all rock. iLO especially rocks, but to leverage the full functionality, you'll need to get the Advanced License, which opens up full blown remote console capabilities (via Java). It's a separate piece of hardware that, as long as the server PSUs have power flowing into them, lets you do things like remotely power on/off/reset the machine (referred to as virtual power), monitor OS crashes (picks up Windows BSoDs and NetWare ABENDShaven't Oopsed Linux on one yet). iLO's allowed me to do everything from BIOS upgrades to fixing NetWare boot issues all from the comfort of my home at 3am in the morning. The build quality is superb...more metal than plastic, so they can weight a bit more, but I expect that of my servers versus desktop boxes. I myself use RAID5 in my DL385 G1's (AMD Opteron), which hold up to six Ultra 320 SCSI drives, on a HP SmartArray controller (64MB of cache thoughneed to upgrade that). The DL320 is a RAID1 on 2x 10k rpm SAS drives, on a...P400 I think w/ 256MB of cache. All battery backed. OS Support is great so far. Just check HP's site for the Proliant Support Packs specific to an OS, as they sometimes provide better drivers than what ships stock w/ the OS (this is especially true for NetWare). --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Al Baker Sent: Thursday, March 27, 2008 8:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! Darren Wright wrote: One of the major reasons we use DL320 / DL380's is the ease of swapping drives, and the integrated ILO BIOS level access.We can support remote sites with ease. If a drive dies we get a notification, a new one is sent and a non-techie can replace it with guidance.No onsite visit. That is worth potentially thousands of dollars. We also leave a recovery CD there that can be inserted if we need to rebuild the system remotely. Never had to, but it's worked in the lab. -D This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Star Wars Echo Sound
That's probably just someone at the NSA snooping your lines and playing tricks on you... g --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rob Schall Sent: Thursday, March 27, 2008 4:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Star Wars Echo Sound We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars type of sound. I expierenced this myself when talking to them. By this, I mean you hear a few words from them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. Has anyone seen this? Thanks, Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider
-Original Message- From: [EMAIL PROTECTED] On Behalf Of Ignacio Ortega A. Let me tell you something i own 200 seats call center, besides have an IT company who develops applications based on asterisk, we are not kid just playing to get some money... so i move millions of min i also resell min to others call centers since 9 months ago now we just open to the public, so yes we bougth the domain a week ago but this not indicates we are trying to steal or someting that`s why we give a free test so if you liked you take it that`s it. Again CDS want to apologise to all users to use this channel to send this message it seems to be a improper channel for that. I would have thought that the Non-Commercial Discussion bit in the To: field of any response to messages on this list would have given away that commercial services being offered here won't be met kindly. And while I can't speak for others here, I've been trained to regard any telephony services offered from the Dominican Republic as meaning something ain't right. Maybe you're an honest guy, but your country's reputation precedes you, unfortunately. Cheers! --J ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min
Piling on... InterNIC says the domain was created almost a week ago, and expires in a year. The registrar is GoDaddy. The owner of the site is located in the Dominican Republic: C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 0 Dominican Republic Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: CDSPORTAL.NET Created on: 14-Mar-08 Expires on: 15-Mar-09 Last Updated on: 14-Mar-08 Administrative Contact: Almonte, Juan [EMAIL PROTECTED] JHALMONTE C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 0 Dominican Republic (809) 220-3278 Judging by the site's purported function, it's nothing more than a front for telemarketers, autodialers, and other ilk of the telephony industry to annoy normal people with. How can you claim five 9's uptime when your domain isn't barely over a week old? Well, I guess if the system hasn't crashed within that first week. But that's hardly a valid measurement, unless you're comparing against Windows Millenium systems. I call scam. --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gonzalo Servat Sent: Friday, March 21, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] www.cdsportal.net wholesale voipprovider --starting at 1.1 cent per min I think this type of abuse is well deserved due to the way he intended to advertise his business, so I'll add a bit of wood to the fire. How about the sign-up form?? Some serious HTML design work going on there. - Gonzalo On Fri, Mar 21, 2008 at 1:15 PM, Tim Nelson [EMAIL PROTECTED] wrote: The template website, page titles, and Gmail contact address surely aren't very convincing. Another crappy VoIP reseller that will fail in a few months taking a handful of customers down... assuming they're legit to begin with. --Tim - Original Message - From: Outback Dingo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 21, 2008 11:06:31 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min My first thought looking at the site was SCAM!!! maybe my second thought would be SCRAM ... is this company even legit On Fri, Mar 21, 2008 at 10:35 PM, Tim Nelson [EMAIL PROTECTED] wrote: Apparently the list description of Non-commercial Discussion isn't clear enough. And now the obligatory beat down: Instant Emergency Response and Delay Free Connection... WOW! I don't even have to call for support because when I have an emergency, response is INSTANT. On top of that... they've also figured out how to eliminate latency!!! Super duper! But wait, theres more!!! They are interconnected with major US carriers like QUEST!!! Not to be confused with QWEST... the little telco company that misspells it's name to differentiate itself from the ULTRA MEGA HUGE telco QUEST. /sarcasm Tim Nelson Systems/Network Support Rockbochs Inc. - Original Message - From: Ignacio Ortega A. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Asterisk-Users@lists.digium.com Sent: Friday, March 21, 2008 10:20:17 AM (GMT-0600) America/Chicago Subject: [asterisk-users] www.cdsportal.net wholesale voip provider --starting at 1.1 cent per min starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pre-pending certain digits (like 9) to an outbound call number
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 '91XX', as the asterisk server
doing faxing junctions into my old Rolm CBX switch, and so I need the '9' digit
to dial outside numbers. However, for deployment, I'd like to save the users
confusion and have the server automatically append that leading '9' digit.
That possible by chance? I assume it is, but off the top of my head, it didn't
seem intuitive. Below is the exten lines for my [fax-out] context, followed by
some test exten lines that wound up failing:
exten = _X.,1,Dial(Zap/g1/${EXTEN}|20)
exten = _X.,n,Busy
exten = _X.,n,Hangup
; Test appending 9?
;;exten = _9XNPANXX,1,Dial(Zap/g1/${EXTEN}|20)
;;exten = _9XNPANXX,n,Busy
;;exten = _9XNPANXX,n,Hangup
I was trying to do some basic matching to the NANP formula to catch when
someone accidentally mistypes a number, but that didn't match up and asterisk
was complaining that no exten lines in the [fax-out] context were matching.
Also, is it possible offhand to block the dialing of certain numbers in the
same context? I.e., just as a check, to block faxes to 900 numbers? I believe
my Rolm CBX will do this for me, as it's got a pretty extensive list of area
codes and exchanges that are known to be sinister in nature pre-loaded
(probably needs updating, though...), but I figured that if I could block it in
asterisk, to do so. Save the Rolm a wee bit of processing and all (it is old,
and probably senile...)
That, and I'd like to filter accidental '9911...' dials using this technique
(which would dial 911 emergency, and that wouldn't be good, since I doubt faxes
are a good method of calling in an emergency (unless they have a color fax and
can discern that the red ink really isn't red ink...)).
Thoughts anyone? Thanks!,
--Josh
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Re: [asterisk-users] Pre-pending certain digits (like 9) to an outbound call number
-Original Message-
From: Gerald A
Try:
exten = _X.,1,Dial(Zap/g1/9${EXTEN}|20)
You want to add the number automagically, not match it being dialled.
Doh!/homer
I knew it'd be simple. I categorically blame Monday for the lapse in my
ability to have seen the obvious :P
Thanks!
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Re: [asterisk-users] NIN Ghosts music (free download) safe for MOH?
-Original Message- From: [EMAIL PROTECTED] On Behalf Of John Faubion Ok now I am curious, if a radio is playing in a store, a restaurant or at the beach, wouldn't that be considered a public performance? And even though the radio station has already paid the license fee, does this mean that the person who owns the radio is also subject to these fees? I know of several key systems with FM radio cards providing MoH and I've often wondered about the ramifications of that setup and the music industry. Well, ASCAP/BMI are stingy on collecting their fees -- one of them even went after a shop for playing the Monday Night Football theme just because they had the TV on that channel when it came on. It seems kind of ruthless if you ask me, but these guys play their cards well, and generally avoid attracting the same the kind of infamy that the RIAA has managed to garner. Unlike the RIAA, they actually pay out their collected royalties. I don't know what their fees even are, but I think they're not too bad from a business' standpoint. Probably can't hurt to call them up and just ask. They might be willing to explain things in better detail. Just be wary if they want your company's information :) --J ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NIN Ghosts music (free download) safe for MOH?
That'd be ASCAP (I think there's another one too). They're the ones known for calling up places, asking to be put on hold to listen to the hold music, then querying on whether it's been licensed or not (among other tactics). Pretty much, unless it's music developed in-house, I wouldn't put it on the hold line unless you're willing to risk a fight with them (and even then, they're likely to make a fuss just for the heck of it). I also suppose this is why Novell support calls are $650 per-incident -- because of the large variety of hold music you'll hear on their support radio. Groups like Metallica, Smashmouth, etc.., even a live DJ. That all adds up --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Latham Sent: Wednesday, March 05, 2008 12:36 PM To: Justin Newman; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] NIN Ghosts music (free download) safe for MOH? I would advise against it right now and contact the artist. The RIAA or the (forgot the name, they charge restaurants and stores for playing music over the PA) would assume that if it is popular one of their members owns it. Other than that I would also assume your primary audience must share your musical taste. On Wed, Mar 5, 2008 at 12:10 PM, Justin Newman [EMAIL PROTECTED] wrote: Is the new NIN Ghosts music (free download) safe for MOH? Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.com */ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring modem pools in Asterisk
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, EM
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
asterisk server (which is only being used for fax purposes as a T1 bridge to
iaxmodem/hylafax).
My initial tests were using the 's' extension, because the other T1 card was
mangling the DTMF signaling due to a hardware incompatibility -- the
replacement card lets things work properly, and asterisk hunts for an exten
line starting with the first digit of my extension group, not 'fax'. Though I
can probably do '_55XX' as my extension definition versus '_X.', since my
extensions are four digits long and there's only 100 of them...
I'm assuming in the provided example, the priority value 'n' simply illustrates
where I need to add 2, 3, 4, etc..., as a quick glance at voip-info.org's
explanation doesn't indicate that 'n' is valid. Not sure what the |20 in the
Dial() function refers to, though...
Thanks for the pointers! A few friends said asterisk config files were
horrific, but they're actually not bad at all, TBH (BIND zone files have far
worse syntax).
--jkinard
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Craig Guy
It should look more like this:
exten = fax,1,Dial(IAX2/iaxmodem1/${NumberCalled}|20)
exten = fax,n,Dial(IAX2/iaxmodem2/${NumberCalled}|20)
exten = fax,n,Dial(IAX2/iaxmodem3/${NumberCalled}|20)
exten = fax,n,Dial(IAX2/iaxmodem4/${NumberCalled}|20)
exten = fax,n,Busy()
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Re: [asterisk-users] Configuring modem pools in Asterisk [WAS: Connecting a Rolm CBX to Asterisk via T1?]
Okay, T1 card issue sorted out. New Lesson: Stay Away from TigerJet chips.
Next up, modem pool -- I wanted to know if the below config looked anywhere
near half-sane for defining in asterisk what is essentially a small pool of
four waiting modems that will handle faxes if another modem is busy:
exten = _X.,1,Dial(IAX2/iaxmodem0/${EXTEN})
exten = _X.,2,Busy
exten = _X.,3,Hangup
exten = _X.,4,Dial(IAX2/iaxmodem1/${EXTEN})
exten = _X.,5,Busy
exten = _X.,6,Hangup
exten = _X.,7,Dial(IAX2/iaxmodem2/${EXTEN})
exten = _X.,8,Busy
exten = _X.,9,Hangup
exten = _X.,10,Dial(IAX2/iaxmodem3/${EXTEN})
exten = _X.,11,Busy
exten = _X.,12,Hangup
This seemed logical, but redundant. I've seen the usage of macro's to condense
stuff like that, but I wasn't sure how to have it auto-determine which modem to
use (i.e., iaxmodem0 through iaxmodem3). In my mind, I'm thinking of this in
the form of a for loop:
for each modem in iaxmodem0..iaxmodem3
is it busy?
Yes: Continue
No: Answer
done
done
Is something like that representable in asterisk-speak?
Also pondering ahead for working on outbound faxing, I'm assuming a [fax-out]
context would be somewhat similar as the above, just a different set of
iaxmodems (4-7)?
Thanks!,
--jkinard
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Re: [asterisk-users] Had it with Dell Garbage
Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very likely, 380's as well). I just learned this the hard way. --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Norman Franke Sent: Tuesday, February 26, 2008 5:27 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Had it with Dell Garbage On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED] wrote: On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote: I've had it with Dell server garbage.They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver.They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I change underwear. [end rant]. Can anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards? HP DL380 is my baby. Thanks, Steve Totaro Ditto. We've been using HPs for a while without problem. I'm currently using a DL380 (a recent quad processor one) and it screams. -Norman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
-Original Message- Joshua, You probably mean a T100P? The single E1/T1 card? Been a few years but I remember seeing the NMI Errors on a HP DL380 (the Intel dual Xeon model). Nah, it's classified as a D110P, although the driver says TE110P. And I checked to make sure I had the onboard jumper rigged for T1 (open), not E1 mode (closed). There's another, unidentified jumper on the board too, but I'm not sure what it's for. Seen this one. Actually now I think of it I've seen those as far back as before Asterisk 0.7 with Dell servers. The DL380 T100P (E1) setup was only used in a test for a couple of months and never taken into production so I don't have any recent data. Hmm, so is it possible this card I have is a prototype that wasn't supposed to ever make it to the market? I've seen it in quite a few online stores. That said, because you've also seen similar issues in a HP Proliant system that's very similar to the DL385 G1, suggests it could be related to something weird HP does in this class of system that this driver and card don't seem to like very much. I've got to run some diagnostics later today on it to verify there's nothing bad with the hardware, but I've also checked for BIOS updates and haven't seen anything new for this system. And I know going through HP's support will be utterly pointless, given their phone people barely speak English anymore. Thanks for the info!, --jkinard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
-Original Message- The D110P is a clone card, which is *not* made/sold/endorsed/etc by Digium. I would suggest getting a newer card, which would not exhibit these types of issues. You will save yourself many headaches in the future. Ah, that would explain quite a bit... Let me guess, Chinese knock-off? I should've known something was a little weird when I kept seeing a green PCI card (TE110P) versus a blue PCI card (D110P). But the chips and other components are all in the same exact spot in the images I found, so I figured it was just a manufacturing curiosity. Bah, that's $300 into the bit bucket. Thanks for the info. Time to go yell very loudly at the person I got this from. Cheers, --jkinard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
-Original Message- Did you get that backwards? All my TE110P cards are blue. I got them from a reputable vendor (voipsupply.com) Yanno, it's hard to tell really. I just took another look at Google, and this: http://hardware4less.net.au/images/te120p_large.png Looks like a Digium card (The asterisk logo is visible), but it could be the TE100P too, of which, OpenVox's ripoff version is also green. I'm probably seeing this card in a lot of the online stores, because even this one looks remarkably close to the TE110P, which does appear to be blue, as is the knockoff I have. I knew the ripoff market was pretty bad and shameless, but until you're actually burned by it, you don't fully realize the lengths they'll go to rip a product off. Really all I need is a simple 1-span T1 card. Nothing fancy and nothing expensive (very limited budget for this project unfortunately). I think I might be able to recover most of what I spent on the D110P, so I'm gonna have to find something that's not too much more expensive (talking $300-$450 range), and I guess, just triple check to make sure it's not another clone card :) --jkinard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
Okay, so I've been toying around on the Rolm side, and still getting nothing. Took another look on Asterisk, finally figured out where the debugging could be enabled on the console, and finding a lot of interesting things. Running 'dmesg' simply shows the entire buffer is flooded with 'PCI Master Aborts', and they appear as soon as the zaptel driver tries to do anything in conjunction with asterisk. Further more, there seems to be a problem with chan_zap and the my_zt_write function, in that it gets a -1 return code, Resource temporarily unavailable (which comprised the bulk of the asterisk debugging output). I've attached a snipped version of that output, notably removing about 2000 lines of chan_zap repeating the Resource temporarily unavailable error. Do I need to look at downgrading asterisk and zaptel to 1.2.x, or might this be some conflict with the Proliant DL385 hardware that currently hosts the T1 Card? FYI, asterisk-1.4.18 and zaptel-1.4.8. Thanks!, --jkinard [Feb 19 13:37:44] DEBUG[15625]: chan_zap.c:1441 zt_enable_ec: Enabled echo cancellation on channel 1 [Feb 19 13:37:44] DEBUG[15625]: pbx.c:1831 pbx_extension_helper: Launching 'Dial' -- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, IAX2/iaxmodem0/s) in new stack [Feb 19 13:37:44] DEBUG[15625]: rtp.c:1585 ast_rtp_make_compatible: Channel 'IAX2/iaxmodem0-1' has no RTP, not doing anything [Feb 19 13:37:44] DEBUG[15625]: channel.c:3277 ast_channel_inherit_variables: Not copying variable TRANSFERCAPABILITY. [Feb 19 13:37:44] DEBUG[15625]: chan_iax2.c:2935 create_addr: prepending 4 to prefs [Feb 19 13:37:44] DEBUG[15625]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/iaxmodem0-1 [Feb 19 13:37:44] DEBUG[15430]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - iaxmodem0-1 [Feb 19 13:37:44] DEBUG[15430]: chan_iax2.c:10432 iax2_devicestate: Checking device state for device iaxmodem0-1 [Feb 19 13:37:44] DEBUG[15625]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel IAX2/iaxmodem0 [Feb 19 13:37:44] DEBUG[15430]: devicestate.c:287 do_state_change: Changing state for IAX2/iaxmodem0-1 - state 4 (Invalid) [Feb 19 13:37:44] DEBUG[15430]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for IAX2 - iaxmodem0 -- Called iaxmodem0/s [Feb 19 13:37:44] DEBUG[15430]: chan_iax2.c:10432 iax2_devicestate: Checking device state for device iaxmodem0 [Feb 19 13:37:44] DEBUG[15430]: chan_iax2.c:10440 iax2_devicestate: iax2_devicestate: Found peer. What's device state of iaxmodem0? addr=16777343, defaddr=0 maxms=0, lastms=0 [Feb 19 13:37:44] DEBUG[15430]: channel.c:1068 channel_find_locked: Avoiding initial deadlock for channel '0x81ba9f0' [Feb 19 13:37:44] DEBUG[15450]: app_queue.c:595 handle_statechange: Device 'IAX2/iaxmodem0-1' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Feb 19 13:37:44] DEBUG[15430]: devicestate.c:287 do_state_change: Changing state for IAX2/iaxmodem0 - state 6 (Ringing) [Feb 19 13:37:44] DEBUG[15450]: app_queue.c:595 handle_statechange: Device 'IAX2/iaxmodem0' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. -- Call accepted by 127.0.0.1 (format ulaw) -- Format for call is ulaw [Feb 19 13:37:44] DEBUG[15442]: channel.c:2763 set_format: Set channel IAX2/iaxmodem0-1 to write format ulaw [Feb 19 13:37:44] DEBUG[15442]: channel.c:2763 set_format: Set channel IAX2/iaxmodem0-1 to read format ulaw -- IAX2/iaxmodem0-1 is ringing [Feb 19 13:37:44] DEBUG[15625]: rtp.c:1502 ast_rtp_early_bridge: Channel 'Zap/1-1' has no RTP, not doing anything [Feb 19 13:37:44] DEBUG[15625]: chan_zap.c:5019 zt_indicate: Requested indication 3 on channel Zap/1-1 [Feb 19 13:37:44] DEBUG[15625]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel Zap/1-1 [Feb 19 13:37:44] DEBUG[15430]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for Zap - 1-1 [Feb 19 13:37:44] DEBUG[15430]: devicestate.c:287 do_state_change: Changing state for Zap/1-1 - state 0 (Unknown) [Feb 19 13:37:44] DEBUG[15450]: app_queue.c:595 handle_statechange: Device 'Zap/1-1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Feb 19 13:37:44] DEBUG[15625]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel Zap/1 [Feb 19 13:37:44] DEBUG[15430]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for Zap - 1 [Feb 19 13:37:44] DEBUG[15430]: devicestate.c:287 do_state_change: Changing state for Zap/1 - state 6 (Ringing) [Feb 19 13:37:44] DEBUG[15450]: app_queue.c:595 handle_statechange: Device 'Zap/1' changed to state '6'
Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
Okay, some more interesting tidbits to throw out incase someone has run into this before. I've found out that th D100P has been EOL'ed by Digium due to it being a bit weird with certain systems, and I suspect my HP Proliant DL385 server may be one of those. Anyone used this card on such a system, running Suse (or at minimum, kernel 2.6.16.5x), and seen either PCI Master Aborts or NMI Errors being fired around in dmesg? I moved the the D110P card out of what was a 100MHz PCI Slot and into a 133MHz PCI slot. The PCI Master Aborts vanish, but get replaced by this: Do you have a strange power saving mode enabled? Uhhuh. NMI received for unknown reason 00 on CPU 1. Uhhuh. NMI received for unknown reason 21 on CPU 0. Dazed and confused, but trying to continue Do you have a strange power saving mode enabled? Dazed and confused, but trying to continue Do you have a strange power saving mode enabled? Ad infinium. The interesting part of all of this? The whole T1-PRI Bridge setup works now. I dial my test extension, and a fax modem comes screaming back at me a few seconds later. That timing oddity with my Rolm system vanished with moving this to the PCI 133MHz slot I guess. The downside is my server locked up a few times until I passed a couple of options given to me by Digium's tech support. I'm going to go run some hardware diagnostics on the server itself to make sure it's not going all emo on me or something, and then see what Digium can maybe help with. But I thought I'd see if anyone's had similar or other odd cases on DL385 hardware. Cheers!, --jkinard -Original Message- Okay, so I've been toying around on the Rolm side, and still getting nothing. Took another look on Asterisk, finally figured out where the debugging could be enabled on the console, and finding a lot of interesting things. Running 'dmesg' simply shows the entire buffer is flooded with 'PCI Master Aborts', and they appear as soon as the zaptel driver tries to do anything in conjunction with asterisk. Further more, there seems to be a problem with chan_zap and the my_zt_write function, in that it gets a -1 return code, Resource temporarily unavailable (which comprised the bulk of the asterisk debugging output). I've attached a snipped version of that output, notably removing about 2000 lines of chan_zap repeating the Resource temporarily unavailable error. Do I need to look at downgrading asterisk and zaptel to 1.2.x, or might this be some conflict with the Proliant DL385 hardware that currently hosts the T1 Card? FYI, asterisk-1.4.18 and zaptel-1.4.8. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restricting registration for peer 'iaxmodem0' to60 seconds
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michelle Dupuis Sent: Tuesday, February 19, 2008 5:50 PM To: 'Asterisk Users List' Subject: [asterisk-users] Restricting registration for peer 'iaxmodem0' to60 seconds 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
Hi all,
So I'm trying to work on this complex fax server setup, and part of it involves
connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on
using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates
HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100%
of the way there. When dialing the fax extension from my Rolm phone, I get
several seconds of silence followed by error tone. But on asterisk's CLI, I
see this:
-- Starting simple switch on 'Zap/2-1'
-- Starting simple switch on 'Zap/3-1'
-- Starting simple switch on 'Zap/4-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/2-1, IAX2/iaxmodem0/s|10|r)
in new stack
-- Called iaxmodem0/s
-- Call accepted by 127.0.0.1 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxmodem0-5 is ringing
-- IAX2/iaxmodem0-5 answered Zap/2-1
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/3-1, IAX2/iaxmodem0/s|10|r)
in new stack
-- Called iaxmodem0/s
[Feb 15 15:40:22] WARNING[24329]: chan_iax2.c:7542 socket_process: Call
rejected by 127.0.0.1: Busy
-- Hungup 'IAX2/iaxmodem0-1'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'Zap/3-1' status is 'CHANUNAVAIL'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, IAX2/iaxmodem0/s|10|r)
in new stack
-- Called iaxmodem0/s
[Feb 15 15:40:30] WARNING[24327]: chan_iax2.c:7542 socket_process: Call
rejected by 127.0.0.1: Busy
-- Hungup 'IAX2/iaxmodem0-3'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL'
-- Hungup 'Zap/3-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, IAX2/iaxmodem0/s|10|r)
in new stack
-- Called iaxmodem0/s
[Feb 15 15:40:35] WARNING[24327]: chan_iax2.c:7542 socket_process: Call
rejected by 127.0.0.1: Busy
-- Hungup 'IAX2/iaxmodem0-4'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'Zap/1-1' status is 'CHANUNAVAIL'
-- Hungup 'Zap/4-1'
The Rolm gives me error tone just before the Starting simple switch messages
begin to appear, so it's almost like the Rolm is not waiting around long enough
for the asterisk server to answer, before it jumps to the next configured T1
channel, runs out of channels (I only configured four in the Rolm and on
asterisk).
Here's my configuration for asterisk. Is anything amiss by chance?
Standard T1
Signalling is EM Wink, 200ms wink time (as far as I can tell)
Mode is ESF and format is B8ZS
/etc/zaptel.conf is:
span=1,1,0,esf,b8zs
em=1-4
loadzone = us
defaultzone=us
/etc/asterisk/zapata.conf is:
[trunkgroups]
[channels]
language=en
context=default
switchtype=national
signalling=em_w
wink=200
channel = 1-4
usecallerid=yes
callerid=asreceived
cidsignalling=bell
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=no
transfer=no
canpark=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
busydetect=yes
busycount=6
faxdetect=incoming
/etc/asterisk/extensions.conf is:
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g0; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0)
[fax-in]
exten = s,1,Dial(IAX2/iaxmodem0/${EXTEN},10,r)
Thoughts?
Thanks!,
--Josh
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Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
-Original Message-
From: Lee Howard
So, okay, there are four calls coming in on the Zap (strange, but...)
There's definitely some kind of a timing error here. I cut my channels back
down to 1, as the Rolm isn't waiting long enough for an answer back from the
asterisk server, and it gives up too early with a busy tone now. What I'm
seeing is the asterisk server taking too long to respond in kind, only to find
the Rolm's quit and gone home already.
Also, asterisk seems to have signalling=em and signalling=em_w mixed up, as I
have to use signalling=em to see a wink sent back down to my Rolm. em_w does
nothing.
An attached text file (rolm-asterisk-chatter.txt) is what my Rolm is seeing.
Notes on each line are on the right and are my additions.
Another attached text file shows what iaxmodem is doing during all of this.
Something about adjusting skew.
Here's what Asterisk itself sees (appears long after the Rolm went to busy
tone):
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, IAX2/iaxmodem0/s) in
new stack
-- Called iaxmodem0/s
-- Call accepted by 127.0.0.1 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxmodem0-3 is ringing
-- IAX2/iaxmodem0-3 answered Zap/1-1
-- Hungup 'IAX2/iaxmodem0-3'
== Spawn extension (fax-in, s, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
And the other calls get busy and improperly run through the auto
fallthrough process (you *need* a Hangup in your dialplan fax-in context).
Added, how does this look?
exten = s,1,Dial(IAX2/iaxmodem0/${EXTEN})
exten = s,2,Busy
exten = s,3,Hangup
I think that your zaptel/zapata configuration between the Rolm and
Asterisk on that T1 is misconfigured. Set it up for PRI if you can...
it'll be a lot easier, is my guess.
Unfortunately, the Rolm only speaks plain T1 talk. It's too old for PRI. We
have an Adtran Atlas unit infront of it that does the PRI-T1 translation that
we get from our carrier, but to get another card for the Adtran is more than
I'll be able to weasel out of my manager for now.
Cheers!,
--Josh
TRK#STATE INL/XDI CODE DIGITS PROCESS TEM SZ
--- -- --- --- --- --
1 IDLEOU
Idle
1 OUTPULS S01/011501 80 OU
fax ext. dialed (80=trk group)
1 OUTPULS S01/011501
1 IDLE
1 IDLE RESZ DELAY
Rolm quits here; busy tone
1 SI RSVD
1 RING-IN
Asterisk rings back
1 DIAL TO R01/011103
1 DIAL TO R01/011103W
Wink sent to Rolm
1 BUSY
Busy because no one answered
1 BUSY
Asterisk hangs up
1 IDLE
1 IDLE RESZ DELAY
1 IDLE
[2008-02-15 17:11:11] Incoming call connected s, , .
[2008-02-15 17:11:12] Answering
[2008-02-15 17:11:12] Adjusting skew to -50.
[2008-02-15 17:11:12] Adjusting skew to -100.
[2008-02-15 17:11:12] Adjusting skew to -150.
[2008-02-15 17:11:12] Adjusting skew to -200.
[2008-02-15 17:11:12] Adjusting skew to -250.
[2008-02-15 17:11:12] Adjusting skew to -300.
[2008-02-15 17:11:12] Adjusting skew to -350.
[2008-02-15 17:11:13] Adjusting skew to -400.
[2008-02-15 17:11:13] Adjusting skew to -450.
[2008-02-15 17:11:13] Adjusting skew to -500.
[2008-02-15 17:11:13] Adjusting skew to -550.
[2008-02-15 17:11:13] Adjusting skew to -600.
[2008-02-15 17:11:13] Adjusting skew to -650.
[2008-02-15 17:11:13] Adjusting skew to -700.
[2008-02-15 17:11:13] Adjusting skew to -750.
[2008-02-15 17:11:14] Adjusting skew to -800.
[2008-02-15 17:11:14] Adjusting skew to -850.
[2008-02-15 17:11:14] Adjusting skew to -900.
[2008-02-15 17:11:14] Adjusting skew to -950.
[2008-02-15 17:11:14] Adjusting skew to -1000.
[2008-02-15 17:11:14] Adjusting skew to -1050.
[2008-02-15 17:11:14] Adjusting skew to -1100.
[2008-02-15 17:11:14] Adjusting skew to -1150.
[2008-02-15 17:11:15] Adjusting skew to -1200.
[2008-02-15 17:11:15] Adjusting skew to -1250.
[2008-02-15 17:11:15] Adjusting skew to -1300.
[2008-02-15 17:11:15] Adjusting skew to -1350.
[2008-02-15 17:11:15] Adjusting skew to -1400.
[2008-02-15 17:11:15] Adjusting skew to -1450.
[2008-02-15 17:11:15] Adjusting skew to -1500.
[2008-02-15 17:11:15] Adjusting skew to -1550.
[2008-02-15 17:11:15] Adjusting skew to -1600.
[2008-02-15 17:11:16] Adjusting skew to -1650.
[2008-02-15 17:11:16] Adjusting skew to -1700.
[2008-02-15 17:11:16] Adjusting skew to -1750.
[2008-02-15 17:11:16] Adjusting skew to -1800.
[2008-02-15 17:11:16]
