Re: [asterisk-users] Server Hardware
I am curious as to what hardware folks are using successfully from HP or DELL. I will likely be running just a quad span T1 card with the system. HP DL380 G4, 4GB mem, 2x 146GB U320 in RAID1, dual hotswap PS HP DL360 G4, 2GB mem, 2x 146GB U320 in RAID1, dual hotswap PS Some Dell models may have issues. Check the Digium website for compatibility (and perhaps the list archives). Both HP boxes work fine with 2 or 4 port E1 cards (hyperthreading is turned off). what are the advantages of turning HT off? btw: i prefer HP servers (above 3xx) because you can do health monitoring really nice (fans, temp, ps status, etc) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Hardware
Keyboardot ragadtam, hogy va'laszoljak Tzafrir Cohen osszedobalt bytejaira: btw: i prefer HP servers (above 3xx) because you can do health monitoring really nice (fans, temp, ps status, etc) configure lm_sensors on just about any system built in the recent years and you'll get those. well, that could be an option, if the mobo is _not_ specialised, for example i could not monitor anything via lm_sensors (or other) stuff on my ibm x235 boxes. and hp also can report prefailure (and then you can ask for replace, as most hardware have prefailure warranty). Btw, i'm not working for HP, but have the best experiences with monitoring them - that not means it's a better box than others. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting an cellphone to asterisk
You can use FXO card to connect gsm adapter with analogue line, (170Eur for used one, and about 350 for new) also you can use bri card, with isdn gsm adapters (about 800Eur for a 2channel) and you can go for junghanns and voismart for a pci card with asterisk support (with sms) junghanns.net has some neat gsm boards that can do this On Aug 20, 2006, at 6:38 PM, Alvaro Cornejo wrote: Hi Is there a way to connect an Cellphone to asterisk in order to route calls though it?. This is what I want to do: Here is much cheaper to call from cell to cell than from fixed line to cell. So I want to connect a cell to the asterisk box and create a rule to route calls to a cell through the cell connected to the asterisk box. Is it possible? Can I do it with the standard data USB cell-pc or I need a special cable/connection? Did someone worked this? Wich cell brand/model can I use for that? Any tips would be appreciate. Regards Alvaro !DSPAM:44e8a400319061757298282! winmail.dat ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:44e8a400319061757298282! -- Roy Sigurd Karlsbakk [EMAIL PROTECTED] (+47) 98013356 --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk VOIP / Mikrotik
have a 10 mb ethernet connection from my ISP into ether1 on a PC - Mikrotik 2.9.23 installed. ether2 is the rest of my network behind the router. How do I prioritize packets such that VOIP calls ALWAYS get a clean channel through to my Asterisk server, which resides behind that router ? Things sound choppy at best at the moment. not the best, but the easiest way is to check queueing, make a queue dedicated (so channel*(80k if g711||30k if g729)) to voip and max the bandwidth of other=all-voip of course there is an option in mikrotik if you want to dig deeper, to match on udp/sip and give much more priority -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 License to Bridge calls through VOIP provider?
Ps. If you know anything about legal issues asked abouta g729 please post it here:) if you are briding g.729, without transcode, and you will NOT stay in mediapath (canreinvite=yes), you don't need g.729 licence -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap DMTF detect error
Hi! Today, as the linux is runnig 136 days ago, with asterisk running 50days ago both * and zaptel is 1.0.10 all the pbx worked well, but they called me at the morning, because the IVR does not detect any DMTF code. (DMTF detect is not worked via sip trunk and dtmfmode=inband with worked with dmtfmode=rfc2833). Any ideas about that? (yes, i know, i have to upgrade :-) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Negotiation
I have two polycom phones. One on a slow link, and one on a fast one. I'm trying to set the phone on the slow link to use G729 as it's first preference, and the phone on the fast link to use G711 as it's first preference. sip.conf has: [general] allow=ulaw allow=g729 [slow-link] ; Override codecs for slow link phone. allow = g729 allow = ulaw When the slow link phone dialls the fast link phone, it sends G729 as it's first preference in the INVITE to Asterisk. Asterisk then sends G729 as the first preference in the INVITE to the fast link phone. Why doesn't Asterisk send G711 instead? This raises an interesting question. If one phone uses G729, and one G711, then Asterisk is going to have to transcode, and I am going to use up a G729 license. It would seem more beneficial for it to work the way it is now. That is, both legs are using G729. Why is this better? It doesn't chew up a G729 license as there is no transcoding, and heck, if one of your call legs is G729, then the G711 party isn't going to hear anything better anyway. Thoughts? don't forget the following: if canreinvite=yes, asterisk will NOT stay in mediapath, so, it going to ask both parties to negotiate codec, and say hello to the stream. (if both parties supports g729, and can negotiate it, no licence will be used) if canreinvite=no, * will STAY in mediapath, so both parties will negotiate with asterisk itself, and will not care about other side. that means, if caller has g729, and callee has g711, asterisk WILL transcode. if both parties have g729, asterisk will NOT transcode, but 2 licence will be used! as i experienced, the codec order in sip.conf [general] will take priority over [user] -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Negotiation
No, we aren't intending to check for available g729 codecs that's why we wanted to have ulaw as a backup when no g729 codecs where available. That won't work. If it's trying to use G729, it will still try even when the licenses are all in use. So you need to either force it g729 and make sure there are always licenses for it available, or use ulaw and make sure there is enough bandwidth. The other option is to write your own code that checks to verify the licenses are free somehow, and then tampers with the codec preferences? I think Brett (trixter) has some ideas/work in this direction already. i heard somewhere, when g729 licences are gone, it will work as g711, is this info FAKE? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM gateway flooded cell - how to detect?
Keyboardot ragadtam, hogy va'laszoljak Colin Anderson osszedobalt bytejaira:
I think, if you should receive network busy, or unreachable (or at least
something, you should handle). You can also try your cellphone, if it gives
better result. Before moving your adapter, you also can try, to buy a
directional
gsm antenna, and direct it to another cell. If you have some time and electrical
knowledge, you can do something like Satellite rotation - so you can force your
gsm gateway to roam to another cell.
here is the status info of my voismart gsm board:
MCC MNC LAC ID BSIC ARFCN RxLev
216 01 0022 3053 2149 -49 dBm
RxLev Sub: -44 dBm
RxLev Full: -40 dBm
RxQual: 0 (BER less than 0.1%)
RxQual Sub: 5 (BER 3.8% = 5.4%)
RxQual Full: 0 (BER less than 0.1%)
Timeslot: 0
TA: 0
RSSI: = -51 dB,
BER: 99 (N/A)
gsm*CLI
Adjacent cells (6)
# MCC MNC LAC ID BSIC ARFCN RxLev
1: 216 01 0022 3144546 -70 dBm
2: 216 01 0022 3054751 -72 dBm
3: 216 01 0022 2eed654 -78 dBm
4: 216 01 0022 2eed656 -87 dBm
5: 216 01 0022 2ef5744 -89 dBm
6: 216 01 0022 2fdc 2043 -90 dBm
as you can see, there is 7 cells (with the 2cm, included rubberduck antenna)
and 4 of them have nice signal.
(I'm going to ask them if there is a chance to manually disable a specified
cell to not to connect)
We are using an Ateus VoiceBlue to GSM gateway calls on our * 1.0.9 server.
It works perfectly fine, except at peak periods, say, 10 AM and 3 PM. At
that point, calls get dropped (not gateway'd) and Asterisk jumps to the next
priority in the dialplan. Our interpretation of this is that the local GSM
cell is flooded with other calls and can't service our request, so nothing
to to with Asterisk or the gateway. No matter how hard we try, during
off-hours, we can't replicate this behavior. My question is how to detect
this behavior and relay the call out to our PRI instead. I've had a couple
of ideas so far, but nothing has panned out:
1. Use the ${DIALSTATUS} variable, however when the condition occurs, the
variable is set to NOANSWER which is the same setting if the guy doesn't
pick up his phone, so it does me no good, since I can't correctly detect
whether it is the gateway or whatever. Maybe an AGI which sets a timer to
detect ringtime? More information: This is different than if the gateway is
full and can't service the request, which I am already successfully testing
for before the dialplan makes the determination to use the gateway in the
first place or not.
2. Dial the target cell using the gateway and the PRI simultaneously, so
this masks the condition. If the gateway kacks, then the call would still go
through the PRI to the target cell. This would work, however I am using the
'r' option to dial, in order to detect early audio if the user has his cell
off to advance the dialplan. When I do this, and the user answers, the PRI
channel gets an early-audio indicator from the GSM provider (The person you
are calling can't answer blablabla ), and Asterisk drops to the next
priority in the dialplan, which I do *not* want to do, until the user has
hung up or doesn't answer. Getting rid of the 'r' option is not in the
cards.
Another idea which just occured to me is to physically move the gateway to
another location a few km away that we have a VPN tunnel to, and just route
calls over there - another cell, maybe not so saturated, right? The danger
there is that the gateway is not on the LAN so the side effect is that our
infrastructure becomes more fragile i.e. if the VPN is down the gateway
doesn't work. Still, I think it's worth a try.
Anybody have any spitballs about how to work around this issue? When the
gateway works, it saves is $2-4K a month in airtime, so I definitely don't
want to abandon it. My GSM provider (Rogers) could care less about working
with me to address this, since it is more revenue for him.
tia
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voismart GSM - no billsecs
I have a Voismart GSM card. I have calls through going fine. But in the cdrs, all the calls have disposiotion of NO ANSWER and the billsecs are 0. I am using Asterisk 1.2.7, visdn 0.16, kernel 2.6.11-12, on CentOs 4.2 that's call received via vgsm interface ,+3620xxx6626,s,gsm417, +3620xxx6626,VGSM/pannon2/1,SIP/800-5d28,Dial,SIP/800,2006-07-20 16:17:24,2006-07-20 16:17:26,2006-07-20 16:18:47,83,81,ANSWERED,DOCUMENTATION that's call placed via vgsm: ,800,0630xxx4904,from-internal,OfficePBX 800,SIP/800-06b1,VGSM/pannon0/1,Dial,VGSM/pannon0/0630xxx4904,2006-07-20 09:52:39,2006-07-20 09:52:58,2006-07-20 09:54:08,89,70,ANSWERED,DOCUMENTATION and this version: http://www.visdn.it/download/snapshots/visdn-devel-20060622.tar.bz2 Thanks to Matteo, everything is going right (also sms in and out) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intel vs amd motherboards
i don't think there is ANY difference with 1 or 2 SATA HDD. however here is my single proc Xeon2.8 (512k) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 6 9 -14 ulaw - 3 - 1 2 2 1 6 9 -14 alaw - 3 1 - 2 2 1 6 9 -14 g726 - 3 2 2 - 2 1 6 9 -14 adpcm - 3 2 2 2 - 1 6 9 -14 slin - 2 1 1 1 1 - 5 8 -13 lpc10 - 4 3 3 3 3 2 -10 -15 g729 - 4 3 3 3 3 2 7 - -15 speex - - - - - - - - - - - ilbc - 4 3 3 3 3 2 710 - - and here is a dual Xeon3.2(1M) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 4 9 -14 ulaw - 2 - 1 2 2 1 4 9 -14 alaw - 2 1 - 2 2 1 4 9 -14 g726 - 2 2 2 - 2 1 4 9 -14 adpcm - 2 2 2 2 - 1 4 9 -14 slin - 1 1 1 1 1 - 3 8 -13 lpc10 - 3 3 3 3 3 2 -10 -15 g729 - 2 2 2 2 2 1 4 - -14 speex - - - - - - - - - - - ilbc - 3 3 3 3 3 2 510 - - the conclusion to me, is comparing transcoding capabilities with show translation is like bogoMIPS... I have recently build 2 machines, one with an Intel Pentium Dual Core CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2 HDDs. Here are the show translations from both: Intel Dual Core machine: pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 517 -17 ulaw - 2 - 1 2 2 1 517 -17 alaw - 2 1 - 2 2 1 517 -17 g726 - 2 2 2 - 2 1 517 -17 adpcm - 2 2 2 2 - 1 517 -17 slin - 1 1 1 1 1 - 416 -16 lpc10 - 3 3 3 3 3 2 -18 -18 g729 - 4 4 4 4 4 3 7 - -19 speex - - - - - - - - - - - ilbc - 3 3 3 3 3 2 618 - - AMD 64 bit machine: pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 313 -12 ulaw - 3 - 1 2 2 1 313 -12 alaw - 3 1 - 2 2 1 313 -12 g726 - 3 2 2 - 2 1 313 -12 adpcm - 3 2 2 2 - 1 313 -12 slin - 2 1 1 1 1 - 212 -11 lpc10 - 3 2 2 2 2 1 -13 -12 g729 - 4 3 3 3 3 2 4 - -13 speex - - - - - - - - - - - ilbc - 4 3 3 3 3 2 414 - - This shows that the AMD 64 bit is worth much more than just the price difference. On 7/6/06, Andrew Kirch [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Don Sent: Wednesday, July 05, 2006 11:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] intel vs amd motherboards If you want to handle, lets say 1000 calls or more at the same time, you should of course use a better processor. In my opinion, it doesn't matter whether you
Re: [asterisk-users] intel vs amd motherboards
As we are talking about pbx boxes (for large office/enterprises/ maybe ITSP?) i think we are using server grade boxes (like hp ml3xx or bigger) I have some servers with fan on cpu heatsink, but most of them are using only heatsink on cpu, and redundant fans. I think, we need some real life comparison to decide, what to choose. i'm not a cpu expert, but who knows, if dual amd is better for transcoding or dual xeon? i think it can as big weight on paralellisation as big weight on horsepower also, don't you think? Another thing, is what to choose? another cpu (so go for dual, or quad) or bigger cache inside? (probably another 3.2G/1M xeon would cost less, than replace the existing with a 3.2G/2M) So i would welcome (and maybe pay for) a real life test what says: AMD opteron will do x paralell alaw-g.729 dual opteron, fx66 and the same for intel pentium extreme, duo core, xeon with 512k cache, xeon with 1M cache, and probably with 2. and also Xeon DP. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding to mobile phone
is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network. [from-gsm] exten = s,1,Dial(Zap/$your_mobile) that's all -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to mobile phone
what brand of gsm gateway do you think works well with asterisk? voismart.it - quadgsm -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving faxes and then sending them on
I'm trying to setup a system where incoming faxes are received using
SpanDSP and then send on to another (remote) fax machine. The SpanDSP
part is working excellently, however I dont seem to be able to get
the forwarding part to work. Heres what I put into my extensions.conf:
exten = s,4,Answer()
exten = s,5,Set(FAXFILE=/tmp/fax-${UNIQUEID}.tif)
exten = s,6,Set([EMAIL PROTECTED])
exten = s,7,Set(EMAILADDR=${ARG1})
exten = s,8,rxfax(${FAXFILE}|debug)
exten = s,9,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} $
{CALLERIDNUM})
exten = s,10,Dial(${ARG2})
exten = s,11,txfax(${FAXFILE}|caller)
exten = s,12,Hangup
Asterisk does start dialing at priority 10 however as soon as the
remote fax hangs up that call gets destroyed as well.
Is there anyway to do something like this?
what about making a callfile?
like s,10,system(make_faxfile ${FAXFILE} ${DST})
this file have to do something like this:
Channel: SIP/trunk/12345
MaxRetries: 1
WaitTime: 20
Application:txfax
Data:/var/spool/asterisk/fax/testfax.tif
than
s,11,system(cp -a faxcallfile /var/spool/asterisk/outgoing/)
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: VGSM Trouble: Kind people, help me please...
Thanks a lot for responding. I did what you recomended, and it works now. At least I can make simple calls out. Did not try the incoming part though. Now it is still unclear : - how to make the Dial application choose the first available channel? the easiest (for you) is installing freepbx (or amportal) set up the trunks (i mean all the four channels) and add all the trunks to outbound routing - or how to get CID out of the interface? Does it set the Global variables as it is in Zaptel? it works right. - ...and a lot of stuff alike, as it usually happens with newly developing project... you R welcome -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callback Application: Suggestions Please.
Keyboardot ragadtam, hogy va'laszoljak Tigran Kocharyan osszedobalt bytejaira: 1. Customer Calls the outgoing number which is a PSTN line connected to my Zap channel 2. Asterisk captures the Caller ID and calls back the customer. 3. As soon as the customer picks up the phone, asterisk plays a promt to enter the Destination number. 4. Asterisk Connects the Outgoing number through another channel (SIP/IAX/ZAP) and bridges the call. 5. After the completion, I should see the Disconnect Reason and the Duration for each leg of the call. The first two steps are quite evident. Now the trick comes on step 3. How to Dial out a number and listen for DTMF tones? After this, maybe park the call, or send it to conference playback(hellomate) DISA(1234|context) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VGSM Trouble: Kind people, help me please...
It was a pain in the ass, to setup the driver first times... however, they just released a stable(?) driver what works well. http://open.voismart.it/ configure the vgsm.conf file as the following: [general] sms_spooler = /usr/lib/asterisk/sms_spooler sms_spooler_pars = -it [global] rx_gain = 255 tx_gain = 255 set_clock = yes [card0] device = /dev/vgsm0s0 pin = context = gsm sms_service_center = +3620930xxx ;operator_selection = auto ;operator_id = that file will make operate the first module (the nearest to the antennas) you should try to run asterisk as user root (asterisk -u root) the stuff installs the files suited for user root and not for user asterisk -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting gateways which time out
set qualify=yes and do chanisavail I would like to know if there is a way to detect gateways which time out (because of network problems or hardware failure for instance) when you send traffic to them. So when you do: Dial(SIP/[EMAIL PROTECTED]) If a call couldn't get through because the gateway has timed out, i want to do something about it. The idea would be to suspend gateway which time out for 60 minutes, and then if calls still don't go through, suspend them permanently and send an email alert. you don't need to do that, just make chanisavail with option j to jump to next gateway if the primary is unavail -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI (I.430) over ethernet
CPE-Asterisk(NT Mode)-ip-Asterisk(CPE)-NT? maybe it will fit for you? if yes, i think you can work with the following budget: via epia board ~85$ mini itx case (small size!) ~85$ ram ~20$ DiskOnChip (or HDD) ~20 - ~50 HFC BRI ~50$ so globally ~300-350/side you can also go for patton something of ~800$ BRI is basic rate ISDN, and consists of 2 channels. They are not the same thing. The redfone is a PRI to TDMoE converter, I'm after something that does the same thing for BRI. Does anyone know of a hardware adapter that can take ISDN BRI frames (I.430) and encapsulate them in Ethernet (any form, but TDMoE would be really cool), in much the same way that the redfone does for PRI? (yes I have asked this before in looser terms, but it was a while ago :) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wav49 size for a 3 minute voicemail
Hi, I tried to find a reference in terms of size but got back a bunch of tech documents and couldn't get the idea of wav49 format. wav49 format is supposed to be half the size of a normal wav right? so, how much disk space takes to save one minute of audio in wav49? I trying to do some capacity planning for a voicemail server. in my experience, it's 1 mbyte/10minutes of call -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI (I.430) over ethernet
CPE-Asterisk(NT Mode)-ip-Asterisk(CPE)-NT? Have you already tried such setup ? What are the benefits of using Asterisk instead a dedicated CPE ? you can extend the range via ethernet/ip? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices of g729 codec
Talk to digium about this on [EMAIL PROTECTED], they might be able to help you out there. Zoa Chris Mason (Lists) wrote: I have no problem with paying Digium the $10 for G729 licenses, everyone has to make money. It's the administration of the licenses that sucks. I experiment with different hardware a lot, and make up demo machines to install for customers with available hardware. I have to put G729 licenses on them, usually $100 each time, and when I install the real hardware for the client, I can't transfer the licenses. If I scrap that machine or change the interfaces, that's a $100 loss. I believe when you buy a number of licenses, that should determine how many instances you can use, regardless of how you want to deploy them. In short, the method of enforcement is poor and leads to resentment from customers. Surely Digium can construct a better system? i think, for those of us, who would like to transfer licences from one box to other (i mean more than 1-2 or 10), we would have to buy a hardware base lock (of course, i don't care about, if the lock would contact digium once a day or so) like usb, or a dumb pci ethernet card, so if we need we can move it to other. what do you think? (sadly there is no a 7day demo licence or anything to test) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk behind cisco pix 506
i try to make asterisk work behind a cisco pix 506. After deactivating the sip fixup i´m able to register but I didn´t hear another party. It´s dialing and connecting but silence. Does anybody has some tips or a sample config for that issue ? allow any to asterisk? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Mini-Box M300
Did anyone try to install Asterisk on the Mini-Box M300 with a Versa mini-ITX board 1GHz VIA x86 CPU? The box looks promissing, but I am not sure if Digium cards are compatible with the mother board (Versa mini-ITX) Also I am not sure if the 1GHz VIA processor can handle a Digium 24 port analog board, or an E1 digital board. If anyone had tried the Mini-Box, the processor, of the mother board, can he please give me a feedback about their use for Asterisk installations. it depends on, if you want to have g.729 (if yes how many)? i tried a pIII/733mhz, with digium 1xPRI. 6 sip-pri calls with alaw codec, worked nice, with 0.02 load on the box. by the way, pIII/933mhz can do 6 paralell g.729-alaw transcode with about 0.6 or 0.8 load (sip-sip calls). so if you are not doing transcode and conference, i'm sure you can drive all the channels paralelly. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P fails to initialize
So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this point, I get the message: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxo dmesg gives me: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.5 Echo Canceller: MG2 Failed to initailize DAA, giving up... wcfxo: probe of :00:0e.0 failed with error -5 this is a problem if the card shares interrupt with something on MOtherBOard. configure the slot to other irq, or move the card to other irq. lspci -v will be your friend. Check if someone uses same IRQ as motorola card. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P fails to initialize
Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.5 Echo Canceller: MG2 Failed to initailize DAA, giving up... wcfxo: probe of :00:0e.0 failed with error -5 these lines means, your x100p is not initialized - therefore cannot be used by zaptel. the problem below, reported by ztconfig. as the usage of zaptel device is following: modprobe zaptel modprobe module_of_card (like wcfxo) if it found, ztcfg if you see the device in /proc/zaptel/1 (or 2 or so) you can start asterisk and enjoy the device. you can believe me, this problem is in relation with irq sharing. (as i meet with that problem every time i have installed more than one card in a box - what i did more than 20 times) Hello. ZT_CHANCONFIG failed on channel 1: No such device or address (6) I think that this error, is saying that its X100P is not connected in slot PCI correctly. He makes a test, he changes the X100P of slot and he sends for the list the commands lspci, dmesg, cat /proc/interrupts. I wait to have helped. Best Regards Josué 2006/6/3, Woodoo People .pGa! [EMAIL PROTECTED]: So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this point, I get the message: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxo dmesg gives me: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.5 Echo Canceller: MG2 Failed to initailize DAA, giving up... wcfxo: probe of :00:0e.0 failed with error -5 this is a problem if the card shares interrupt with something on MOtherBOard. configure the slot to other irq, or move the card to other irq. lspci -v will be your friend. Check if someone uses same IRQ as motorola card. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@ RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] Compiling chan_bluetooth
does chan_bluetooth working well now? (integrating sound and signal channels in BT?) If yes, it's better than using a cheap GSM adapter (like 150Euro)? ps: i have tested it in last year with nokia6310, but with no luck. Just to close the thread. The problem was that I was using an old version of the code. If anyone has the same problem, you can download the code from here: http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz Good luck, Danko - Original Message - From: Danko Miocevic [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 30, 2006 8:48 PM Subject: Re: [Asterisk-Users] Compiling chan_bluetooth I´ve found a solution to my problem, I forgot to install the posix development libraries.. now the error has dissapeared to make place to a new error! :D I still can´t compile. The new error says: cc: chan_gsm_bt.o: no se usó el fichero de entrada del enlazador porque no se hizo enlace cc: -lbluetooth: no se usó el fichero de entrada del enlazador porque no se hizo enlace It says that the file wasn´t used because the linker didn´t make link... don´t know what to do.. Any ideas? Danko - Original Message - From: Danko Miocevic [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 27, 2006 12:51 PM Subject: [Asterisk-Users] Compiling chan_bluetooth Hello, I´m trying to use my phone with asterisk to get GSM connectivity but I can´t compile the code. I got the asterisk, chan_bluetooth, zaptel and libpri sources, the last two ones compiled perfectly. I have added this to the /usr/src/asterisk/channels/Makefile: include /usr/src/chan_bluetooth/Makefile and I´ve also added chan_bluetooth.so to the CHANNEL_LIBS var. When I do make install in the asterisk directory I get lots of this error: /usr/src/chan_bluetooth/chan_bluetooth.c:2469: error: dereferencing pointer to incomplete type and some others like: /usr/src/chan_bluetooth/chan_bluetooth.c: En la función `remove_sdp_records': /usr/src/chan_bluetooth/chan_bluetooth.c:2937: error: `sdp_session_t' undeclared (first use in this function) /usr/src/chan_bluetooth/chan_bluetooth.c:2937: error: `sdp' undeclared (first use in this function) /usr/src/chan_bluetooth/chan_bluetooth.c:2938: error: `sdp_list_t' undeclared (first use in this function) /usr/src/chan_bluetooth/chan_bluetooth.c:2938: error: `attr' undeclared (first use in this function) /usr/src/chan_bluetooth/chan_bluetooth.c:2939: error: `sdp_record_t' undeclared (first use in this function) /usr/src/chan_bluetooth/chan_bluetooth.c:2939: error: `rec' undeclared (first use in this function) /usr/src/chan_bluetooth/chan_bluetooth.c:2948: aviso: implicit declaration of function `sdp_connect' /usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `BDADDR_ANY' undeclared (first use in this function) /usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `BDADDR_LOCAL' undeclared (first use in this function) /usr/src/chan_bluetooth/chan_bluetooth.c:2948: error: `SDP_RETRY_IF_BUSY' undeclared (first use in this function) I´m working on a stable debian, kernel 2.6 and my asterisk is 1.2. I really don´t know what is happening, if someone has an idea I´d be glad to hear it. Thanks for reading, Danko ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP voice recorder
I believe that Cisco does the monitoring/recording that way. We've been working with a company that has implemented Cisco's approach and they are having problems with the recording due to network design (eg, high- availability dual-everything. Port mirroring is only picking up half the conversation). Their recording method apparently works when it can see both sides of the conversation. Don't know anything about their software for that function however. this mirror port problem can be solved if you connect the asterisk box thru the monitor-box what's operating in bridge mode. that also make the possibility (if you have a 2port nic) to connect asterisk directly the switch via another cable, using Spanning Tree with higher cost. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I guess my server capacity is ok
Which DSP based boards does Asterisk support for G729 and are any of these more cost effective than piling on Pentiums? There are none at this time. BTW: Can AMD CPUs handle a higher G729 load in 64 bit mode? Yes. The G.729 codec we distribute is marginally (6-7%) faster on AMD64 in 64-bit mode than in 32-bit mode. what about xeon processors in 64bit mode? as i know the 3.2GHz processors with 1M cache and above are supporting 64bit operations. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk restarting in a minute
yes, it was a typo... and the problem of working too much... crontab? I restart my asterisk nightly with cron but a simple typo could make that every minute instead of every day... shrug Probably any of you meet with the following problem: asterisk is restarting in a minute (if no active call) if active call, it says cannot receive a call due to restart in progress. even if i starting with -c, i have no disconnected, but see the stuff restarting. i've tried to recompile, older version, virgin config, etc. same results. it's happened after a power loss, on a ext3 fs, sitting on a raid1. astdb was deleted, log is not showing any interesting things. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dealing with trafication tone
Hi! Any of you played with tarification tone? We are planning to insert and asterisk box in front of a panasonic with PRI, but the old pbx still needs the tarification tone. Btw, it would be nice, if we could use the tone is asterisk itself (rather than connect the cdr with a tarification system). Thanks! -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] connecting asterisk to pstn help
look for SER and Asterisk on voip-info. I think, you plan to got to UA-SER-(mediaproxy)-Asterisk-PSTN if yes, ser will communicate UA (user agent) on one leg, and asterisk on other. you can use your asterisk to billing and pstn connection. on incoming call dial $phone/ip.address.of.ser Here i going explain what Iam doing and where i need help .. Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients ,SER is running on port 5060 and Asterisk on 5065, here i need to forward pstn calls to asterisk and i am planning to connect asterisk to a Cisco Gateway , when sip client calls to pstn SER will recieve invite message and it forwards to asterisk 1)how the Asterisk will handle this call with rtp 2)and when pstn customer calls the call goes in to SER and it looks the 'location' database and it will reject call because it is not registerd user so, we take pstn call directly to asterisk and we forward call from asterisk to SER and i want to know is how the SER handle this call that means when SER found a sip client it invites that sip client and which mediaproxy is going to handle this call the SER's or Asterisk's Can we use only one mediaproxy for both SER and ASTERISK by loading modules in ASTERISK so that it will be easy for billing ..??? please explain me how the process will take here bcoz i am with lots of questions and confusions in this particular process hope some body will solve my headache confusion ..Thanks in advance Kindly regards, Ravi. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I can't call PSTN numbers
autocreatepeer=yes [ser_box1] type=peer username=ser_box1 insecure=yes canreinvite=no context=from-internal host=ip.address.of.box nat=yes disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm It doesn't work for me :-( How do you have the peer configuration in asterisk, to connect ot SER? exten = _4XX,1,Dial(SIP/[EMAIL PROTECTED]) it works to me (my provider sends me the last 3 digits) I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called [EMAIL PROTECTED] SIP/SER_ip_address-ec75 is circuit-busy Everyone is busy/congested at this time (1:0/1/0) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk restarting in a minute
Hi! Probably any of you meet with the following problem: asterisk is restarting in a minute (if no active call) if active call, it says cannot receive a call due to restart in progress. even if i starting with -c, i have no disconnected, but see the stuff restarting. i've tried to recompile, older version, virgin config, etc. same results. it's happened after a power loss, on a ext3 fs, sitting on a raid1. astdb was deleted, log is not showing any interesting things. any ideas please? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Panasonic PBX
The place I currently work at has a Panasonic Key system with 9 extensions, and no voicemail. It services 2 PSTN lines. I am hoping to use Asterisk to host voicemail (I would like to use the IVR also, but I don't even know if or how it would work). Do I need to use a PRI between the two, or is there a simple solution? I would like people to be able to answer the phone and transfer the call to voicemail if the person is not there, or after so many rings, it goes right to voicemail. I'm not sure what is needed? I have seen the integration How-To but that requires the PRI, and wasn't sure if that was the ONLY way to go. i don't think it's the only way. there is no logical difference between Zap/g0 (includes channel 1-15 of pri) than Zap/g0 (includes bri1-1 and bri1-2 to bri4-1 and bri4-2 if using bristuff) or NTPorts (includes port1,2,3,4 using mISDN) keep in mind, the cheapest FXS is a port ATA but if call-status is a must, and more than 8 channels, go for PRI -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I can't call PSTN numbers
exten = _4XX,1,Dial(SIP/[EMAIL PROTECTED]) it works to me (my provider sends me the last 3 digits) I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called [EMAIL PROTECTED] SIP/SER_ip_address-ec75 is circuit-busy Everyone is busy/congested at this time (1:0/1/0) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER qualify
Hi! I know that is not SER discuss, but probably some of you faced with the same problem: to detect trunk status (ok/unreachable) in *, it's a must, to set qualify=yes as * connecting to SER, it's not replying to qualify messages, so even i can use it well without qualify, with qualify it's says unreachable immediately. What i have to set in SER to reply? Thanks! -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy Signals
I think asterisk dropping you to s-BUSY, s-CONGESTED, s-UNREACHABLE
priority, better have a look there
(you can play a busy tone, or playback(called-party-is-busy))
A few employees have noticed some problem here and there when trying to
make outgoing phone calls. After it happens, they try again, and are
able to call through.
The dial plan for outbound calling looks like below. Which I know they
are getting to the Congestion part (which explains the busy) but what I
can't seem to figure out is the cause for why they are getting sent there.
exten = s,1,SetCallerID(${ARG1})
exten = s,2,Wait(2)
exten = s,3,Dial(${TRUNK1}/${ARG2})
exten = s,4,Congestion(10)
exten = s,104,Congestion(10)
The log for a call looked like this
May 26 12:21:08 VERBOSE[6997] logger.c: -- Channel 0/4, span 1 got
hangup request
May 26 12:21:08 VERBOSE[16613] logger.c: -- Zap/4-1 is circuit-busy
May 26 12:21:08 VERBOSE[16613] logger.c: -- Hungup 'Zap/4-1'
May 26 12:21:08 VERBOSE[16613] logger.c: == Everyone is busy/congested
at this time (1:0/1/0)
My question is it asterisk having an issue with the PRI or is the PRI
really reporting the number is busy. I know one case like this I was
calling home, and which when I got through to them, they were not even
on the phone. Are there any tests that I can run on the T1 card in the
server to the PRI? Any suggestions would be helpful.
Kevin
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mISDN FAX
I have mISDN installed and working correctly but I am unable to receive a fax through the connection. Is there anything I need to do to get this running? I have nvfaxdetect installed but I think this only works with sip, iax and zaptel i have working fax in and out. as i have direct fax number, fax detect is not a problem, but as i know, faxdetect only support by zap channels natively however, write a mail to mISDN developer, probably they can improve their driver. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soekris hadware
i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here. My question are the following: personally i have no experience, but i think you have to forget g.729, and also handling more than 2-3 paralell calls. 1) does the Digium TDM400P fit in the soekris box with a 4801 SBC or a bigger box is needed? Any suggestions about where to pick up another box? 2) does the Digium TDM100P (already discontinued) fits fine in a soekris box? 3) running asterisk in a soekris 4801 SBC, what is the perfomance related to sip connections, analogue call quality and both mixed at the same time? 4) what is the actual state of fax support into asterisk / [EMAIL PROTECTED] ? it's working for me in and out. 5) It's posible to create personalized dialplans that enables a hidden or passcode/password protected menu for remote administration or remote use of the pbx? yes -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which is the best fax-modem for testing ?
Keyboardot ragadtam, hogy va'laszoljak Rich Adamson osszedobalt bytejaira: I agree with most of the points, however i have installed several systems with x100p and/or hfc based ISDN, and voip trunk. If they user don't forget to use the configured prefix for using pstn for fax, everything is nice. Even if we are using digium fxo/fxs ports, or using ATA (Linksys pap2 or Sipura) on FXS side. I'm not saying that is a 100% solution, but it works for SOHO i think. Which fax-modem would you pick if you had to test fax capabilities ? For instance, before releasing a new PBX system offering fax connectivity, you would like to make sure you comply with most fax machines and protocols. As you can't afford you buy and maintain tens of such fax machines nor can't afford to test by hand each protocol, it's tempting to buy an all-inclusive fax-modem and run a program instead. Which one would you choose for that ? The fax modem is not really the issue with asterisk. By far, the majority of existing analog fax machines installed and being sold today will function just fine with asterisk. If you sell an asterisk system into an analog pstn environment, any fax machine will function through asterisk if you use the Sangoma A200D analog card with fxo and fxs modules. (Very stable and very reliable fax transmissions.) If you sell an asterisk system into a digital pstn environment (eg, PRI), any fax machine will work with Sangoma or Digium digital cards, however the fxs interface to the fax machine may be very questionable in terms of reliability and usability. If you sell an asterisk system with external pstn gateways (eg, ATA adapters), better be careful as the majority of inexpensive gateways will not function reliably with an analog fax machine. If you're thinking T.38 fax capability, forget it for now. Some folks were working on adding T.38 support into asterisk, but its not in stable code as yet to the best of my knowledge. Also, according to Steve Underwood, T.38 implementations in current fax machines are of questionable quality. If you're thinking in terms of high volume faxing, then look towards the hylafax (or whatever) approach. If you're thinking in terms of faxing via VoIP providers, reliability will be less then acceptable if you get it to work at all. Bottom line: the most reliable method of integrating fax support into an asterisk system today (without implementing hylafax or whatever) is through the use of the Sangoma A200D analog card, as it keeps the pcm data flow on the card (fxs - fxo); and, removes the impact that pci bus, shared interrupts, system applications, ethernet dropped packets or jitter, ATA issues, and other disruptive elements from the analog fax data path. If you search the list archives for the past two years, you'll find a couple of point solutions other then mentioned above that do work, but most of them are dependent on some specific element (eg, full moon) that cannot be reliably replicated in every asterisk installation. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Adapter
I am seeking for the SIP Adapter which is providing the dual FXs ports. I can get some in the market, did some one experience that using Zyxel P-2002 ATA compatible with Asterisk? Further more, does Auto-Provisioning ATA useful to work with Asterisk? Please advice, Good experience ATA is needed. Sipura2100 is well ok for that money (about $100) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confused !
Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?
I've got a HFC ISDN card that I'm using with chan_misdn and it basically behaves like crap. Echo is waaay worst then echo I get TDM400 card, sound is choppy (there other side is allays complaining about sound interruptions) and to top it all it detects fake DTMF's all the time. Is this a chan_misdn problem or is it a card problem? I really need to get this fix and I need to know the way to go. I don't want to throw money at a better card if the card is not the issue but if that's the only solution, I'll need to order the card ASAP! i'm using 1port (billion bipac), quad and octoBRI cards from beronet. all of them working nice, beronet recommend to use kernel 2.6.12+ and asterisk 1.2.x and also newest misdn-mqueue from www.beronet.com -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rxfax problem
Hi! Anyone meet with the following problem? May 12 15:51:44 WARNING[14399] channel.c: Unable to find a codec translation path from ulaw to unknown May 12 15:51:44 WARNING[14399] app_txfax.c: Unable to restore read format on 'SIP/neopost1-8083' May 12 15:51:44 WARNING[14399] channel.c: Unable to find a codec translation path from ulaw to unknown May 12 15:51:44 WARNING[14399] app_txfax.c: Unable to restore write format on 'SIP/neopost1-8083' May 12 15:51:44 DEBUG[14399] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record. May 12 15:51:44 NOTICE[14399] pbx_spool.c: Call completed to SIP/neopost1/0676505921 May 12 15:51:44 DEBUG[14420] app_rxfax.c: == May 12 15:51:44 DEBUG[14420] app_rxfax.c: Fax successfully received. May 12 15:51:44 DEBUG[14420] app_rxfax.c: Remote station id: May 12 15:51:44 DEBUG[14420] app_rxfax.c: Local station id: May 12 15:51:44 DEBUG[14420] app_rxfax.c: Pages transferred: 1 May 12 15:51:44 DEBUG[14420] app_rxfax.c: Image resolution: 7700 x 3850 May 12 15:51:44 DEBUG[14420] app_rxfax.c: Transfer Rate: 9600 May 12 15:51:44 DEBUG[14420] app_rxfax.c: == May 12 15:51:44 WARNING[14420] channel.c: Unable to find a codec translation path from alaw to unknown May 12 15:51:44 WARNING[14420] app_rxfax.c: Unable to restore read format on 'mISDN/2-1' May 12 15:51:44 WARNING[14420] channel.c: Unable to find a codec translation path from alaw to unknown May 12 15:51:44 WARNING[14420] app_rxfax.c: Unable to restore write format on 'mISDN/2-1' May 12 15:51:44 VERBOSE[14420] logger.c: == Spawn extension (ext-fax, in_fax, 5) exited non-zero on 'mISDN/2-1' May 12 15:51:44 VERBOSE[14420] logger.c: -- Executing Hangup(mISDN/2-1, ) in new stack May 12 15:51:44 VERBOSE[14420] logger.c: == Spawn extension (ext-fax, h, 1) exited non-zero on 'mISDN/2-1' or any idea how to overcome? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] email - fax gateway with billing possibilities?
does anyone have an idea how it could be possible to do email - fax gatewaying with asterisk + app_txfax, but still keep track of who sent the fax? i've thought a little about smtp auth, but it doesn't look too easy to integrate smoothly with asterisk i don't know what your problem is. ask the user to use a callerID as a sender ([EMAIL PROTECTED]) or pair his sender id to callerid, than do the billing on the callerid. that's my .02 -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One sided call
Hi! I found, that there is 4 options for nat: -no -never -yes -always no and never is ok but sometimes yes, and sometimes always worked for me :-o I am having problem diagnosing a call problem. On both a Cisco phone and a Linksys 942 I am only getting one side of the call when connected over a WAN link or internet connection. I have set nat=yes and qualify in sip.conf and the phone registers fine. I can hear the other end, but they do not hear anything, no voice or dtmf. I found a tip about changing the RTP rate from .03 to .02 on Sipura phones to match Asterisk rate and did that. I also made sure the RTP range for the phone and the server was set to 1 thru 2. These phones work fine when on the same subnet as the server. The server shows the following message: -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P vs Kudzu, original was: Problems with TDM400P and FXO modules
If ztcfg -v shows your card it's working, it's OK. probably kudzu doesn't that your card is already configured I am sorry cause i post this questions is not related to your problem, but i am having problem detecting my TDM400P which is a TDM400P problem. I manage to installed the card with compiling with zaptel and it got 2FXS and 2FXO. I am having problem while reboot or restart the system. Kudzu seem to detect TDM400P network card which is Tiger Jet Network Modem /ISDN hardware removed. Is quiet confusing cause if i remove the hardware configuration is still working and something the card no circuits flow. I am using centos 4.2, asterisk , zaptel, libpri 1.2.2 and runnning freepbx as interface to asterisk. When i run ztcfg -v , output as below : Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. Thanks in advance. On 5/9/06, Rich Adamson [EMAIL PROTECTED] wrote: Bogdan Tocu wrote: The outline is like this : Board 1 channels 1-4 # channels 1-4 Port 1 unused Port 2 unused Port 3 - FXO module - not working Port 4 - FXO module - working ok Board 2 - 4 FXO modules - all working ok # channels 5-8 Board 3 - 4 FXO modules - none works #channels 9-12 Any ideeas? Which part of the previous post did you not understand? You wrote the entries in /etc/zaptel.conf assuming that what you are calling Board 1 really is Board 1, and its not. Its Board 3 using your numbering scheme. I don't know of any way to determine exactly how three identical boards are numbered, so you'll have to experiment to determine which board holds channels 1-4, which has channels 5-8, and which has channels 9-12. Its obvious from the error message that you posted that channel 9 and 10 correspond to what you are calling Board 1 (since there are no modules in the first two positions of that board). So, change your /etc/zaptel.conf and zapata.conf to address the empty channel 9 and 10 slots. Once you get asterisk to run, then (and only then) you can place a call to each pstn line and see which Zap channel corresponds to which board by watching the CLI. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN and Asterisk
Straight cable for TE mode and Xover for NT mode you have to make a call, than L1 should go up I have a Cologne Chip Designs GmbH ISDN network controller and I want to terminate voip calls via this ISDN card. My question is: How I must to wire the ISDN equipment with my ISDN card? With normal cable or crossover? How I can to check if ISDN card is linked with ISDN equipment? In this moment I have 1:1 cable between ISDN's, the mISDN is installed and misdn show stacks said Stack Addr:4104 Port 1 Type TE Prot. PMP L2Link UP L1Link:DOWN Thanks and sorry for my english. PS is my first experience with ISDN lines. -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 vs Panasonic IP Hybrid
Hi! I want to make a call to/from Panasonic IP pbx thru asterisk via H323. H323 is working nice, I can call/receive using netmeeting -- Executing Dial(OH323/[EMAIL PROTECTED], OH323/[EMAIL PROTECTED]|15|tr) in new stack -- H.323 call to [EMAIL PROTECTED] with codec(s) alaw -- Outbound H.323 call to destination '[EMAIL PROTECTED]', channel 'OH323/[EMAIL PROTECTED]'. -- Called [EMAIL PROTECTED] -- H.323 call 'ip$localhost/15803-305e28d0' cleared, reason 24 (Call ended with Q.931 cause [28 - Invalid number format]) -- Hungup 'OH323/[EMAIL PROTECTED]' if Panasonic 'd like to dial, i've got cause [111 on it anyone meet with this? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN and Asterisk
On Wednesday 10 May 2006 16:02, Woodoo People .pGa! wrote: Straight cable for TE mode and Xover for NT mode you have to make a call, than L1 should go up I connected the card to ISDN provider's equipment via crossover cable, but portinfo report: Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX What is not useable for PBX, the ISDN line or my ISDN card? hfcpci is loaded with this command: modprobe hfcpci layermask=0x3 protocol=0x32 type=0x01 if misdn show stacks displays something, usually the driver install is successful. as i have said TE (Terminal Equipment) mode you have to use straight cable, and if your card is NT (and another is TE) you have to use isdn cross cable (3 to 6 and 4 to 5) to connect with. So generally use straight cable to the 'wall' (Network Terminal) and cross cable (or jumper to NT mode if your card supports) to another Equipment (like PBX) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on EM64T
I'm looking to install Asterisk on an EM64T Dell 1850. PERC raid 1, 1GB ram, single 3Ghz Xeon. Any red flags or anything I should know? Should I bother installing a 64 bit OS? (gentoo-amd64)? Does asterisk work in 64 bit mode? Should I turn hyper threading off? Etc? I'm doing audioconferencing with Asterisk meetme using a SIP channel on a DL360 (dual EM64T). I'm running Debian Etch for amd64. amd64 optimized system is really better for Xeon than i386? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best CPU (of expansion hardware?) for g.729 enc/dec ?
Hi! What do you think (or benchmarked) what would be the best for g.729 encoding on a mediagateway running asterisk? we would plan to encode alaw(g711) to g.729 and probably zap(pri)-alaw to g729 do i have to go for dual opteron instead of xeons? do 1M cache and 2M cache CPU worth the money? what about powerPC? thanks for any ideas -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk/Zaptel 64-bit?
Thank for your reply and advice, Patrick. The reason I didn't add a RAID1 to the server was that I am not sure if RAID1 on SATA II is stable on FC5 yet and also since the HDD isn't hotswap (at least i don't think SATA HDD can hotswap). There will always be a downtime for me. I don't know the stability of software RAID1 with SATA either. In spite of having to take the box down to replace one of the broken disks, you can do it at a time that suits you/your customers best. I think you can also add a third disk that will act as a hot spare. there are already exist SATA hotswap cages, so hotswapping is not a problem if you are using a hotswap compatible controller. If you are thinking in low budget, i recommend whole spare server :-) if not, try something with spare everything :-) (disk,ps,ram,cpu) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2.x with app_rxfax
Does anyone has a working one? mine always receives the fax, but then cannot set back the format and goes away :-( PLease help! -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns GSM card
Does anyone have some experience with junghanns GSM cards? I want to know if I can use this cards to send SMS directly from Asterisk box. They look terrible to me. From the picture on their website it looks like you need one antenna per GSM channel (most gateways use 1 antenna per 4 ports). Plus it would suck to have to open your computer to swap the SIM cards... they offer a SIM 'expander' for 2sim/channel that is cabled to back side of the PC Apparently the current market pricing for VoIP - GSM gateways is around 500 euros per port. Now if you can beat this using their hardware, they might be onto something. as i know, it's 900Euro/2port and 1600Euro/4port. voismart.it offers almost same for 1200Euro/4port -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns GSM card
Can anyone tell me if it is possible to send the SMS through this card directly from extensions.conf with some application that takes the text string and converts it to SMS and which colaborates with junghanns card. i think yes, and sure for voismart -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_rxfax problem on 1.2.6
Hi! I'm using freepbx, with * 1.2.6, everything is working nice, except fax handling. the incoming faxes got received: May 6 23:24:39 DEBUG[12505] app_rxfax.c: == May 6 23:24:39 DEBUG[12505] app_rxfax.c: Pages transferred: 1 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Image size: 1728 x 1116 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Image resolution7700 x 3850 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Transfer Rate: 9600 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Bad rows3 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Longest bad row run 3 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Compression type2 May 6 23:24:39 DEBUG[12505] app_rxfax.c: Image size (bytes) 0 May 6 23:24:39 DEBUG[12505] app_rxfax.c: == May 6 23:24:42 DEBUG[12505] app_rxfax.c: == May 6 23:24:42 DEBUG[12505] app_rxfax.c: Fax successfully received. May 6 23:24:42 DEBUG[12505] app_rxfax.c: Remote station id: T-Info2001 May 6 23:24:42 DEBUG[12505] app_rxfax.c: Local station id: May 6 23:24:42 DEBUG[12505] app_rxfax.c: Pages transferred: 1 May 6 23:24:42 DEBUG[12505] app_rxfax.c: Image resolution: 7700 x 3850 May 6 23:24:42 DEBUG[12505] app_rxfax.c: Transfer Rate: 9600 May 6 23:24:42 DEBUG[12505] app_rxfax.c: == May 6 23:24:42 WARNING[12505] app_rxfax.c: Unable to restore read format on 'mISDN/2-1' May 6 23:24:42 WARNING[12505] app_rxfax.c: Unable to restore write format on 'mISDN/2-1' May 6 23:24:42 VERBOSE[12505] logger.c: == Spawn extension (macro-faxreceive, s, 3) exited non-zero on 'mISDN/2-1' in macro 'faxreceive' May 6 23:24:42 VERBOSE[12505] logger.c: == Spawn extension (macro-faxreceive, s, 3) exited non-zero on 'mISDN/2-1' but as seen, macro-faxreceive exiting before converting the fax. I think it's becaus unable to restore format, but not sure. The strange is, same freepbx (amp) works nice on 1.0.x I have installed 1.2.6 from source, but spandsp and appfax got from package. I've tried to install newest spandsp, and appfax, but appfax got failed on compilation. Anyone knows the problem? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk hardware
Keyboardot ragadtam, hogy va'laszoljak Koopmann, Jan-Peter osszedobalt bytejaira: Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-) I'm sorry if i recommend some foolish (i've just joined the maillist) but have you tried PA168 chip based hardphones and ATAs? www.areadfox.com for example -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk hardware
Well, to tell the truth, the phones, what available in Hungary, is 90% working. The other 10% is sometimes bad as you get out off the box, sometimes it's noisy, echoing, crappy sound, rebooting, etc. Is i asked so many folks on Cebit (who resells this phone) most of them, told me, there are two kind of this phone. One is cheap and crappy, other is not as cheap but at least working :-) Either way, i think the chip itself is working nice, has gpl source (look on voip-info.org) so you can go for it, and don't undertake the chip because of a nasty manufacturer. I'd rather shoot myself in the head! other day we had a site that flashed the PA168 chipset phones with new firmware and they all ended up with the same MAC address!! I thought that shouldn't happen normally ... And talk about nasty cheap effects, sidetone, distortion and the list goes on. Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-) I'm sorry if i recommend some foolish (i've just joined the maillist) but have you tried PA168 chip based hardphones and ATAs? www.areadfox.com for example -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk hardware
Since when do these use IAX? He asked for IAX hardphones... If I am mistaken let me know since I am looking for good reliable SNOM-like IAX phones as well! :-) I'm sorry if i recommend some foolish (i've just joined the maillist) but have you tried PA168 chip based hardphones and ATAs? www.areadfox.com for example I believe you mean http://www.aredfox.com/eindex.htm . yes :-) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
