Re: [Asterisk-Users] Codec Selection
On Sun, Feb 05, 2006 at 05:51:40PM +0300, [EMAIL PROTECTED] wrote: > Hi, > > I guess what you mean by a Carrier as Trunk. > > If you have an SIP Trunk i feel the preference list will do the needful. > > > disallow=all > allow=g723 Some clarification here: If you dial using: Dial([EMAIL PROTECTED]) You can use peer-specific codec setting in the specific settings for that peer. If you dial using: Dial([EMAIL PROTECTED]) you use the default settings. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] French and German translations?
Hi, Are there good (and complete, also for the voicemailmain application) french and german translations available for Asterisk 1.2? -- Philippe Lang, Ing. Dipl. EPFL Attik System rte de la Fonderie 2 1700 Fribourg Switzerland http://www.attiksystem.ch Tel: +41 (26) 422 13 75 Fax: +41 (26) 422 13 76 Email:[EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ --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] Billing inbound calls per minute
Michiel van Baak wrote:
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the "Connect fee"(if I put one)
and keeps it that way no matter how long
the call is ...( if no "Connect fee" -stays empty).
i.e.
[inbound]
exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten => 1122334455,3,Hangup
DeadAGI is for hungup channels, not for active channels.
That might be a problem.
Try this:
exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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RE : RE : [Asterisk-Users] Codec Selection
You will need to buy licences for the codec you want, I don't know for
g723, but g729 costs 10$ by channel.
Cheers
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoyé : dimanche 5 février 2006 21:50
À : asterisk-users@lists.digium.com
Objet : RE : [Asterisk-Users] Codec Selection
Hi,
Is there any special configuration for transcoding on
asterisk? Or Asterisk will do it automatically?
---
Olivier Taylor
Sun, 05 Feb 2006 11:51:51 -0800
Hi,
Just forget to choose the Codec on asterisk :(
Only solution is :
Disallow=all
Allow=YourCodec
If client doesn't have that codec you will need to
transcode on asterisk.
If client has that codec,asterisk will do pass-thru
and it will work.
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoyé : dimanche 5 février 2006 20:00
À : asterisk-users@lists.digium.com
Objet : Re: [Asterisk-Users] Codec Selection
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI->get_variable("DIALSTATUS");
if ($discr == "CONGESTION" || $discr == "NOANSWER" ||
$discr == "CHANUNAVAIL")
{
my $dialstr = "$gwtype/$gwip/" . $dialednum . "|30|tTL(" .
($crdeit*1000) .":7000:5000)";
$AGI->exec('Dial', $dialstr);
$discr = "";
}
Any idea?
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
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Re: [Asterisk-Users] Do we need a QOS switch ?
> >> We have 10 people on our network and each person will have a SIP phone > >> connected to our Asterisk server. All phones, Asterisk, other servers and > >> users workstations will be using the same network. The question is: would > >> I need a QOS device to give SIP traffic a chance? Our internal network is > >> 100M. We will have a ISDN30 for outgoing calls. No calls will be made > >> over the internet. > >> We have dealt with this issue in small offices by using phones that contain a switch (Polycom IP500s) and do their own QoS. In other words, all the users' PCs are hooked into their phone, so any excessive traffic does not interfere with the phone. Since the phones then hook directly into the same switch that the PBX (Asterisk) hangs off, quality has been fine. Keep in mind this is for small offices like you describe. Provided the topology of your switches is OK, you should be fine. Just don't uplink to another switch where you can create a non-QoSd bottleneck link. -Ron ___ --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] Hunting for DIDs in Kenya/Nigeria
On 1/25/06, Chris Bagnall <[EMAIL PROTECTED]> wrote: Don't know if anyone's got experiences on this they'd be able to share...I'm trying to obtain numbers for Kenya/Nigeria, but I'm struggling to find acompany selling them. There's one for Nigeria listed on didx.com, but$22/month seems a little steep. Has anyone had any luck getting DIDs forcountries in Africa, and are there any companies out there selling them?Thanks in advance.Regards,Chris --C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons Hi, I'm based in Nairobi, Kenya and can assist. Please contact me offline. Rgds, Alphonse ___ --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] TE411P Really Bad Echo
Stagg Shelton wrote: I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I 1 to 2 _seconds_? There is no echo canceler anywhere that will handle that much echo delay. Did you actually remove the VPM, or just disable it? Please check whether you have this problem with the VPM installed but disabled, because it could be a bad VPM. Finally... I know it's Sunday night, but you should really pursue this with Digium Support tomorrow morning :-) ___ --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] TE411P Really Bad Echo
I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right now, and we have a tdm400p with 4 fxs modules installed as well. If anyone has experience with this card, can you tell me if I am missing something. zaptel.conf = span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #bchan=25-47 #dchan=48 #bchan=49-71 #dchan=72 #bchan=73-95 #dchan=96 fxoks=97-100 loadzone = us defaultzone=us zapata.conf = context=from-pstn switchtype=national pridialplan=national signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=3.0 txgain=0.0 group=0 channel=>1-23 ;;[1103] signalling=fxo_ks record_out=Adhoc record_in=Adhoc [EMAIL PROTECTED] echotraining=800 echocancelwhenbridged=no echocancel=yes context=from-internal callprogress=no callerid=device <1103> busydetect=no busycount=7 channel=>98 ;;[111] signalling=fxo_ks record_out=Adhoc record_in=Adhoc [EMAIL PROTECTED] echotraining=800 echocancelwhenbridged=no echocancel=yes context=from-internal callprogress=no callerid=device <111> busydetect=no busycount=7 channel=>97 Thanks Stagg www.oneringnetworks.com ___ --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] AVAYA H.323 IP phone account and Asterisk
HiI've a softphone account to a AVAYA H.323 system, basically, it has a numeric ID (which is the extension number) and a numeric password. Instead of using the default AVAYA softphone (H.323), can I make asterisk as a H.323 client and login to the AVAYA system via any one of its h323 modules? ___ --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: Re: [Asterisk-Users] IP PAX gateway to PSTN
> >> > >> Thanks for the answers. I really appreciate that. It may be better > for > >> me to talk to local Telco for further price negotiation. > >> > > Going through a VOIP termination service is also good for testing. > > > When going thru a VoIP termination service, do I also need to have a IP > PAX gateway? To not answer your question, but send you in the right direction: If you use a terminaton service, you do not need a phone card. PaulH ___ --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: Re: [Asterisk-Users] IP PAX gateway to PSTN
>Hi, > >I m still not quite understand why I need E1/T1 PRI line with VoIP calls >to normal phone line. I thought I can send calls out thru a normal home >telphone line. If this is the case, I will just need to pay each VoIP call >to phone line at 20 - 30 cents / call. Then you will need to buy an analogue phone card. PaulH ___ --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] IP PAX gateway to PSTN
>> >There are lots of Asterisk users here in Australia...and it's not > illegal. >> > >> >You will probably have to discuss charging with your Telco. >> > >> >PaulH >> > >> >_ >> > >> > >> >> Thanks for the answers. I really appreciate that. It may be better for >> me to talk to local Telco for further price negotiation. >> > Going through a VOIP termination service is also good for testing. > When going thru a VoIP termination service, do I also need to have a IP PAX gateway? Sam > There are quite a few here in Australia. > > PaulH > ___ --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 laptop connected to POTS line
I thought they stopped selling the spa3000 ? --- Damon Estep <[EMAIL PROTECTED]> wrote: > Sipura SPA-3000 will give you 1 fxs and 1 fxo so you > can connect to a > POTS line AND a analog phone at the same time with > one small box. > > Makes a great demo system. > > > -Original Message- > > From: [EMAIL PROTECTED] > [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Dovid > Bender > > Sent: Thursday, February 02, 2006 6:20 AM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] Asterisk on laptop > connected to POTS line > > > > Anyone know of any equipment that I can use to > connect > > a laptop running asterisk to a POTS line (RJ11) ? > > > > Regards, > > Dovid > > > > ___ > --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 > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] IP PAX gateway to PSTN
>> > >> > How much your telco is going to charge you for the PSTN calls depends >> > on your arrangement with the telco... Usually, with proper volume >> > interconnects (say you order a PRI line), these calls are charged per >> > second. >> > >> Do I really need PRI T1 line when I initially setup VoIP network? > > How do you want to send calls out onto the public phone network? > (and here in Australia, we run E1, not T1) > Hi, I m still not quite understand why I need E1/T1 PRI line with VoIP calls to normal phone line. I thought I can send calls out thru a normal home telphone line. If this is the case, I will just need to pay each VoIP call to phone line at 20 - 30 cents / call. Thanks Sam > PaulH > ___ > ___ --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] Strata DK280 + [EMAIL PROTECTED]
We have a Toshiba Strata DK280 and are about to implement it with a server running [EMAIL PROTECTED] 2.4. We have the Asterisk server running fine with a Digium analogue card and it receives and makes calls absolutely fine, of course this is just one line only so only one call at a time. We would like to know the basic steps to running our Strata PBX system with the Asterisk server. It currently uses ISDN technology to handle calls to extensions using a series of cards as a 30 channel system (models of cards available if needed). I assume of course we’ll need an ISDN card in the Asterisk server to be able to ‘talk’ to the Strata system but that’s about where it stops. I would really appreciate it if you could provide some steps on what to do to connect the Strata system to the Asterisk server, what cards / equipment is required to allow the Asterisk server to manage the 30 channels and anything else I’ve missed here. Thanks in advance. Chris Peacock Technical Director Reality Solutions Limited Burnett Buildings West Carr Lane Kingston upon Hull East Yorkshire HU7 0AW Phone: 01482 828000 Fax: 01482 832869 Email: [EMAIL PROTECTED] Website: www.realitysolutions.co.uk Reality Solutions is an ICT provider established in 1995 to service the ever expanding business needs of the local business community. We have over 10 years of experience in delivering IT and Sage solutions for your computing demands including installation, configuration, training, programming (including customisation work for Sage business solutions) and comprehensive on-site support for all hardware and software. Any views or opinions expressed in this email message and attachments thereto are those of the original sender. Nothing contained in this email, its main text and the attachments thereto, are intended to be an offer, proposal, or a suggestion whatsoever to commit Reality Solutions to enter into any agreement, contact, or transaction to purchase, sale, or part with any of its rights. If you are not the intended recipient of this message or a person responsible for delivering it to the addressee, you are hereby notified that you must not disseminate, copy, use, distribute, publish or take any action in connection therewith, as the message might be containing information that is confidential and restricted, thus, subject to legal restrictions and sanctions. ___ --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
[Solved] [Asterisk-Users] Routing Calls via chan_capi with AVM FritzCard
Hello asterisk-users, Christian Schmidt, 04.02.2006 (d.m.y): > Armin Schindler, 04.02.2006 (d.m.y): > > > You really should update to new chan_capi-cm version (you can find it > > on sourceforge.net). > > OK, I gave that a try. > Now, my server is running asterisk 1.0.10 with chan_capi-cm from > SourceForge. > > When calling asterisk from my phone, it rings and rings and rings. > > Asterisk says: > *CLI> == 3413: Incoming call '0012341234' -> '' > Urgent handler > == 3413: CAPI Hangingup > Urgent handler > The "CAPI Hangingup" occurs between the second and the third ring. > > It seems to me as if asterisk doesn't receive a "destination msn". > Unfortunately, I can only access the asterisk server but not the PBX > that provides my ISDN channels... My asterix now accepts calls coming in via CAPI. I had to change my dialplan for the specific context to: [isdn_incoming] exten => s,1,SetLanguage(de) exten => s,2,Dial(SIP/10,60) exten => s,3,Voicemail2(u10) exten => s,4,Hangup exten => s,103,Voicemail2(b10) exten => s,104,Hangup I tried first with "3413" (the MSN) instead of "s". Thanks to all of you who were so kind to answer my postings! Regards, Christian Schmidt -- Durch das Fernsehen wird kein Mensch, aber wahrscheinlich viel Geist getötet. -- Alfred Biolek ___ --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] 1 ISDN BRI to IAX2/SIP... (*) best tool or?...
I have a question, I have to provide a solution for an office that will be almost abandoned, and there will be one or sometimes two persons 2 days a week. The main number however should be preserved. They have several ISDN BRI connections, most of which will be dropped. Only one will be retained, for 2 reasons: 1) It has the ADSL link 2) The number has been the main contact number for over 20 years. What we are looking for is to put a single SIP phone in the office, and have it connect back to an (*) server in the central office, where all other servers are located as well. In the remote office a single machine should be placed to terminate the BRI connection and relay it to the (*) server in the central office. That way the old number can be retained and an active phone can pick up the line as necessary. The preferred protocol to use would be IAX2, obviously. My question is whether there are any tools better suited for this than an old banger (AMD 800 MHz) PC with a HFC-PCI card and (*) relaying (switch) the incoming calls to the central box. (No intelligence there, no AGI scripts, just encode and transmit. Also no phones would need to be logged in to that machine, and outbound calling would only take place in very rare cases when the lines *and* VOIP connections at the central site are all congested...) TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Billing inbound calls per minute
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
> Hi,
> Does anyone have a neat idea as how to
> bill inbound calls per minute(second) real time?
>
> I've been pplaying with astcc, but while
> 'billseconds' stays empty, 'billcost' has
> strange behavior - either stays ampty
> or takes ONCE the "Connect fee"(if I put one)
> and keeps it that way no matter how long
> the call is ...( if no "Connect fee" -stays empty).
>
> i.e.
> [inbound]
> exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
> exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
> exten => 1122334455,3,Hangup
DeadAGI is for hungup channels, not for active channels.
That might be a problem.
Try this:
exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D
"Why is it drug addicts and computer afficionados are both called users?"
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RE: [Asterisk-Users] (newby) Asterisk on the open internet & security
Thanks for the idea, it will work for my two interconnected PBX’s. Happily those two can really be locked down and properly firewalled because they’re on fixed IP’s (allmost). Will using encription require some extra bandwidth? How much? Better yet, if you want some added security you can get IAX2 to do encryption between two asterisk endpoints instead and avoid the extra latency of a vpn layer. Tim. http://www.westhawk.co.uk/ ___ --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] (newby) Asterisk on the open internet & security
> Hey, > > We are running asterisk on the internet, allowing sip phones > at customers locations/laptops etc login and use the calls. > Just make sure to disallow sip users/peers without valid > user/secret in the extensions.conf > (something like this in sip.conf) > [general] > context = sip-default > (and in extensions.conf) > [sip-default] > exten => s,1,Hangup() So this trick allows an anonymous connection onto the * and next it closes the connection (Hangup). Isn't it possible to make Asterisk completely reject a connection if no credentials can be accepted? (Is Hangup() technically the same considering Asterisk uses UDP for SIP?) > If you dont trust and fear someone is sniffing your udp > packets that hold user/secret, you can always setup openvpn > (or whatever vpn solution) and use that to connect first and > tunnel your sip traffic through it Yep, this is an other problem. I might after all allow connections from unrecognized sip phones go to my operator (mabe they're clients!), but sending "clear text" passwords over udp packets is not nice at all. As with other things in life, I don't think anyone's actually actively tracking my moves and trying to hack into my network, but I am afraid of "IT hooligans" detecting my UDP packet on it's way from my home to my office and hacking it just to prove it's possible. Trying to find my own way through this maze I came across this page: http://www.voip-info.org/wiki-SIP+Authentication ...and I ask: What kind of authentication does Asterisk provide with SIP? Is it digest or basic? If it's digest - it's fine with me. If it's basic - I'll have to set up some more "barriers" for calls coming over the public network (like asking for a password from the IVR, before allowing any kind of outgoing calls). I will not be using any kind of VPN because of the extra bandwidth required. > -- > Michiel van Baak > http://michiel.vanbaak.info > [EMAIL PROTECTED] > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D > > "Why is it drug addicts and computer afficionados are both called 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
[Asterisk-Users] Billing inbound calls per minute
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the "Connect fee"(if I put one)
and keeps it that way no matter how long
the call is ...( if no "Connect fee" -stays empty).
i.e.
[inbound]
exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten => 1122334455,3,Hangup
Thanks in advance,
benchev
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Re: [Asterisk-Users] (newby) Asterisk on the open internet & security
On 5 Feb 2006, at 21:11, Michiel van Baak wrote:On 22:38, Sun 05 Feb 06, Cosmin Prund wrote: Hello everyone. I'm again bothering you with a bit of a problem, hopefullynot really a problem. I just need someone to tell me this is ok :-)I'm planning on having two * machines on the open internet (ie: not behind aNAT) and having them talk to each other using IAX2. I can handle all thefire walling requirements in this case easy because at least one of the *'shas a fixed address and I'll be able to filter traffic on IP.If you dont trust and fear someone is sniffing your udppackets that hold user/secret, you can always setup openvpn(or whatever vpn solution) and use that to connect first andtunnel your sip traffic through itBetter yet, if you want some added security you can get IAX2 to do encryption between two asteriskendpoints instead and avoid the extra latency of a vpn layer.Tim. http://www.westhawk.co.uk/ ___ --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: RE : [Asterisk-Users] Codec Selection
On 12:50, Sun 05 Feb 06, Abdul Lateef wrote: > > Hi, > > Is there any special configuration for transcoding on > asterisk? Or Asterisk will do it automatically? If the codecs from both ends are known to asterisk, * will do it automagically for you :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called 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] (newby) Asterisk on the open internet & security
On 22:38, Sun 05 Feb 06, Cosmin Prund wrote: > > Hello everyone. I'm again bothering you with a bit of a problem, hopefully > not really a problem. I just need someone to tell me this is ok :-) > > I'm planning on having two * machines on the open internet (ie: not behind a > NAT) and having them talk to each other using IAX2. I can handle all the > fire walling requirements in this case easy because at least one of the *'s > has a fixed address and I'll be able to filter traffic on IP. > > It's all fine and safe so far. But then it hit me: I'll also want to "log > on" to my business's PBX from home, in order to gain access to some of its > low-rate gateways! That will not work if my office * filters on IP! Nor > would I be able to use a soft SIP phone on my laptop when I'm not at the > office! > > So my question: > > Is Asterisk's built-in security enough? If ALL my sip peers have propper > usernames and secrets set up and my box has only the required ports open, is > it safe to run Asterisk on the open internet? Does anyone run Asterisk like > that? > > I can allmost answer my own question: "You may safely run Asterisk like that > - there are lots of VoIP services providing PSTN termination that way" but, > being new to all this stuff, I'll stay on the safe side and ask. > > Thanks. Hey, We are running asterisk on the internet, allowing sip phones at customers locations/laptops etc login and use the calls. Just make sure to disallow sip users/peers without valid user/secret in the extensions.conf (something like this in sip.conf) [general] context = sip-default (and in extensions.conf) [sip-default] exten => s,1,Hangup() If you dont trust and fear someone is sniffing your udp packets that hold user/secret, you can always setup openvpn (or whatever vpn solution) and use that to connect first and tunnel your sip traffic through it -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called 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] Do we need a QOS switch ?
> Hi, > > We have 10 people on our network and each person will have a SIP phone > connected to our Asterisk server. All phones, Asterisk, other servers and > users workstations will be using the same network. The question is: would > I need a QOS device to give SIP traffic a chance? Our internal network is > 100M. We will have a ISDN30 for outgoing calls. No calls will be made > over the internet. > As long as the current infrastructure is decent, you should be fine without a separate voice switch. PaulH ___ --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] IP PAX gateway to PSTN
> >There are lots of Asterisk users here in Australia...and it's not illegal. > > > >You will probably have to discuss charging with your Telco. > > > >PaulH > > > >_ > > > > > > Thanks for the answers. I really appreciate that. It may be better for > me to talk to local Telco for further price negotiation. > Going through a VOIP termination service is also good for testing. There are quite a few here in Australia. PaulH ___ --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] IP PAX gateway to PSTN
> > > > How much your telco is going to charge you for the PSTN calls depends > > on your arrangement with the telco... Usually, with proper volume > > interconnects (say you order a PRI line), these calls are charged per > > second. > > > Do I really need PRI T1 line when I initially setup VoIP network? How do you want to send calls out onto the public phone network? (and here in Australia, we run E1, not T1) PaulH ___ --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 Selection
Hi,
Is there any special configuration for transcoding on
asterisk? Or Asterisk will do it automatically?
---
Olivier Taylor
Sun, 05 Feb 2006 11:51:51 -0800
Hi,
Just forget to choose the Codec on asterisk :(
Only solution is :
Disallow=all
Allow=YourCodec
If client doesn't have that codec you will need to
transcode on asterisk.
If client has that codec,asterisk will do pass-thru
and it will work.
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoyé : dimanche 5 février 2006 20:00
À : asterisk-users@lists.digium.com
Objet : Re: [Asterisk-Users] Codec Selection
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI->get_variable("DIALSTATUS");
if ($discr == "CONGESTION" || $discr == "NOANSWER" ||
$discr == "CHANUNAVAIL")
{
my $dialstr = "$gwtype/$gwip/" . $dialednum .
"|30|tTL(" . ($crdeit*1000) .":7000:5000)";
$AGI->exec('Dial', $dialstr);
$discr = "";
}
Any idea?
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
__
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[Asterisk-Users] (newby) Asterisk on the open internet & security
Hello everyone. I'm again bothering you with a bit of a problem, hopefully not really a problem. I just need someone to tell me this is ok :-) I'm planning on having two * machines on the open internet (ie: not behind a NAT) and having them talk to each other using IAX2. I can handle all the fire walling requirements in this case easy because at least one of the *'s has a fixed address and I'll be able to filter traffic on IP. It's all fine and safe so far. But then it hit me: I'll also want to "log on" to my business's PBX from home, in order to gain access to some of its low-rate gateways! That will not work if my office * filters on IP! Nor would I be able to use a soft SIP phone on my laptop when I'm not at the office! So my question: Is Asterisk's built-in security enough? If ALL my sip peers have propper usernames and secrets set up and my box has only the required ports open, is it safe to run Asterisk on the open internet? Does anyone run Asterisk like that? I can allmost answer my own question: "You may safely run Asterisk like that - there are lots of VoIP services providing PSTN termination that way" but, being new to all this stuff, I'll stay on the safe side and ask. Thanks. ___ --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] Do we need a QOS switch ?
stoffell wrote: On 2/5/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: We have 10 people on our network and each person will have a SIP phone connected to our Asterisk server. All phones, Asterisk, other servers and users workstations will be using the same network. The question is: would I need a QOS device to give SIP traffic a chance? Our internal network is 100M. We will have a ISDN30 for outgoing calls. No calls will be made over the internet. If you don't overload your internal network, you'll be fine.. Ah... THERE is the key phrase we were looking for. The proposed VOIP traffic will have little impact on the usability of their network FOR VOIP traffic. It is all the other stuff that runs across their LAN that make make VOIP "a really cappy idea", if the don't take steps to ensure that the VOIP traffic is managed properly. With the paucity of details provide by the OP, it is impossible to say, with any degree of credibility, that the "...will be fine..." Do those 10 phone sit on the desks of graphic designers, whose file and print traffic can bring a 100 Mbps segment to its knees? ___ --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 Selection
Hi,
Just forget to choose the Codec on asterisk :(
Only solution is :
Disallow=all
Allow=YourCodec
If client doesn't have that codec you will need to transcode on asterisk.
If client has that codec,asterisk will do pass-thru and it will work.
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoyé : dimanche 5 février 2006 20:00
À : asterisk-users@lists.digium.com
Objet : Re: [Asterisk-Users] Codec Selection
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI->get_variable("DIALSTATUS");
if ($discr == "CONGESTION" || $discr == "NOANSWER" ||
$discr == "CHANUNAVAIL")
{
my $dialstr = "$gwtype/$gwip/" . $dialednum .
"|30|tTL(" . ($crdeit*1000) .":7000:5000)";
$AGI->exec('Dial', $dialstr);
$discr = "";
}
Any idea?
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
__
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RE: [Asterisk-Users] re: questions about sip requests to asterisk 1.2
you need to setup a asterisk peer at port 5070 in sip.conf to get the callreplying correctly to ser.
Cheick Zerbo
Corbimas.com
[EMAIL PROTECTED]
From: Yair Hakak <[EMAIL PROTECTED]>Reply-To: Asterisk Users Mailing List - Non-Commercial DiscussionTo: Asterisk Users List Subject: [Asterisk-Users] re: questions about sip requests to asterisk 1.2Date: Sun, 5 Feb 2006 14:55:32 +0200
hi all,
I keep asking the question and getting no replies, so i'll keep asking :-)
In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically
rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to [EMAIL PROTECTED], it will make it [EMAIL PROTECTED]:5070, and asterisk will match it to 151 in the dialplan.
In asterisk 1.2 asterisk completely ignores the request (even at most verbose level) and an ngrep shows a "not found" returned to SER.
anyone have any idea why this is happening, bug/feature, or how to get it to work the way it did in 1.09? I want to upgrade but I don't want to lose this functionality.
thanks for any help,
yair
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Re: [Asterisk-Users] Codec Selection
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI->get_variable("DIALSTATUS");
if ($discr == "CONGESTION" || $discr == "NOANSWER" ||
$discr == "CHANUNAVAIL")
{
my $dialstr = "$gwtype/$gwip/" . $dialednum .
"|30|tTL(" . ($crdeit*1000) .":7000:5000)";
$AGI->exec('Dial', $dialstr);
$discr = "";
}
Any idea?
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
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Re: [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amount
well I've heard that there are "open source" IP phones given away for free in WALMART, I'm seriously thinking to get couple of 'em!! Truely/ Joe Tahan From: "Brian J. Murrell" <[EMAIL PROTECTED]>Reply-To: Asterisk Users Mailing List - Non-Commercial DiscussionTo: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] Re: delaying "answer" for a number of ringsor an amountDate: Sun, 05 Feb 2006 09:49:08 -0500>On Sun, 2006-02-05 at 05:28 -0600, Joseph Tanner wrote:> >> > Again, give everyone in your home/office a phone connected to asterisk> > (whether it's a sip/iax phone, or a regular phone connected to an ATA,> > or what have you).>>Sure. Wanna send me some ATAs or even IP phones?>>It's all about budget dude. Not everyone has the $$ to outfit the whole>house with IP and IP phones right away.>>b.>>-->My other computer is your Microsoft Windows server.>>Brian J. Murrell ><< signature.asc >> >___>--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 Share a single photo or an entire slide show right inside your e-mail with MSN Premium. Join now and get the first two months FREE* ___ --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] Do we need a QOS switch ?
On 2/5/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > We have 10 people on our network and each person will have a SIP phone > connected to our Asterisk server. All phones, Asterisk, other servers and > users workstations will be using the same network. The question is: would > I need a QOS device to give SIP traffic a chance? Our internal network is > 100M. We will have a ISDN30 for outgoing calls. No calls will be made > over the internet. If you have a fairly decent 100Mbit switch, you'll be fine. I assume you will make up to 10 simultaneous calls, so you can calculate the bandwidth you'll be using. (http://www.voip-info.org/wiki/view/Bandwidth+consumption) When using the G.711 codec, it'll be about 1.5-2Mbps when doing 10 simultaneous calls. If you don't overload your internal network, you'll be fine.. cheers ___ --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] Do we need a QOS switch ?
Hi, We have 10 people on our network and each person will have a SIP phone connected to our Asterisk server. All phones, Asterisk, other servers and users workstations will be using the same network. The question is: would I need a QOS device to give SIP traffic a chance? Our internal network is 100M. We will have a ISDN30 for outgoing calls. No calls will be made over the internet. Thank you in advance! Phil ___ --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 Selection
Hi, I guess what you mean by a Carrier as Trunk. If you have an SIP Trunk i feel the preference list will do the needful. disallow=all allow=g723 Dan On 05/02/06, Abdul Lateef <[EMAIL PROTECTED]> wrote: Hi All,I have one Carrier which is supporting only G.723.1,how i can put in my extentions.conf to send calls tothis GW using G.723.1, because for Clients i canspecify the codec from sip.conf but i am littleconfiuse how i can give specific codec for carriers.your ideas will be appriciated. Yours,Abdul LateefComputer ProgrammerHATIF COMMob: +974 - 5405022ICQ: 276994704MSN: [EMAIL PROTECTED]GoogleTalk: [EMAIL PROTECTED]YM!: abdul_zuDoha Qatarhttp://www.hatif.com__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com___--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
Re: [Asterisk-Users] Re: delaying "answer" for a number of rings or an amount
On Sun, 2006-02-05 at 05:28 -0600, Joseph Tanner wrote: > > Again, give everyone in your home/office a phone connected to asterisk > (whether it's a sip/iax phone, or a regular phone connected to an ATA, > or what have you). Sure. Wanna send me some ATAs or even IP phones? It's all about budget dude. Not everyone has the $$ to outfit the whole house with IP and IP phones right away. b. -- My other computer is your Microsoft Windows server. Brian J. Murrell signature.asc Description: This is a digitally signed message part ___ --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] inform the agent about the queue he is answering
On 2/4/06, nik600 <[EMAIL PROTECTED]> wrote: > On 2/3/06, nik600 <[EMAIL PROTECTED]> wrote: > > On 2/3/06, Script Head <[EMAIL PROTECTED]> wrote: > > > Yes, it is possible. You need to track the queue log and channels via > > > manager console or by tailing logs in real time and then match the > > > destination of the caller by the callerid. Then make the decision which > > > URL > > > to redirect the caller too. None of this comes with Asterisk but it is > > > possible to build. > > hi > i'm trying to tailing logs, this is the problem: > > 1139045971|1139045971.14|700|NONE|ENTERQUEUE||101 > 1139045978|1139045971.14|700|Local/[EMAIL PROTECTED],1|CONNECT|7 > > in the first row you can see that the extension 101 is entered in the queue > 700 > now, when the agents answer from extension > Local/[EMAIL PROTECTED],1 the log reports the second row, but i > need this information first than the answer of the agents! > > can i enable something in the queue logs due to see something like this? > > first log: > 1139045971|1139045971.14|700|NONE|ENTERQUEUE||101 > second log: > ... . .. . . . . | 700 | ringing on 102 | 101 > third log: > 1139045978|1139045971.14|700|Local/[EMAIL PROTECTED],1|CONNECT|7 > > thanks > due to the above problem i'm trying to catch the information with the Manager API...when i'm logged in, what command shall i use to know from what queue is ringing an extension? thanks ___ --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: questions about sip requests to asterisk 1.2
Hi Jean-Michel, have you tried upgrading? can you confirm this behavior? It seems to me this is a major issue for those of us running SER + asterisk, and who dont want to configure each SIP client in SER and asterisk separately. -yair On 2/5/06, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote: > In asterisk 1.2 asterisk completely ignores the request (even at most> verbose level) and an ngrep shows a "not found" returned to SER. >> anyone have any idea why this is happening, bug/feature, or how to get> it to work the way it did in 1.09? I want to upgrade but I don't want> to lose this functionality.Since I use 1.0.9 and use exactly the same scheme, I am interested onhow to upgrade as well.Cheers,Jean-Michel.--Jean-Michel Hiver - http://ykoz.net/Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE___--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
Re: [Asterisk-Users] Search for Links for "Communicating PC to PC in the same lan through Asterisk "
On Sun, Feb 05, 2006 at 01:23:05PM +, John Joseph wrote: > > --- Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > > On Sun, Feb 05, 2006 at 07:51:33AM +, John > > > > > > Here are some relevant keywords: > > > > you basically need to install softphones (software > > VoIP "phones") on the > > computers that use either SIP or IAX. then set up in > > Asterisk extensions > > for both of them (SIP or IAX, depends on the type of > > the softphone). > > > > > > Hi > Thanks for the advice , I have one Linux machine > and another Windows XP machine , If I install > SipXphone from http://www.sipfoundry.org/sipXphone/ > on XP machine will it be fine Sure. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Search for Links for "Communicating PC to PC in the same lan through Asterisk "
--- Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Sun, Feb 05, 2006 at 07:51:33AM +, John > > > Here are some relevant keywords: > > you basically need to install softphones (software > VoIP "phones") on the > computers that use either SIP or IAX. then set up in > Asterisk extensions > for both of them (SIP or IAX, depends on the type of > the softphone). > > Hi Thanks for the advice , I have one Linux machine and another Windows XP machine , If I install SipXphone from http://www.sipfoundry.org/sipXphone/ on XP machine will it be fine Thanks Joseph ___ Yahoo! Photos NEW, now offering a quality print service from just 8p a photo http://uk.photos.yahoo.com ___ --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: questions about sip requests to asterisk 1.2
In asterisk 1.2 asterisk completely ignores the request (even at most verbose level) and an ngrep shows a "not found" returned to SER. anyone have any idea why this is happening, bug/feature, or how to get it to work the way it did in 1.09? I want to upgrade but I don't want to lose this functionality. Since I use 1.0.9 and use exactly the same scheme, I am interested on how to upgrade as well. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] Sirrix PC140 Quad card
Hi,
I have just installed a Sirrux PC140 card for the first time. Managed
to build the drivers and
get * to load them on FC4, but it does not work.
It seems that layer1 in the ISDN is not even activated. The same ISDN
lines connected to a Samsung DCS works so it is not the lines.
I am including my sirrix.conf and the output of some of the * Srx
commands below. Any pointers would be appreciated.
Thank you
Marnus van Niekerk
--
; global settings
[Global]
internationalprefix = 09
nationalprefix = 0
;crypto_app = /root/cvs/sirrix-pci/crypto/crypt
; external link
[out]
mode = TE
ptp = no
context = incoming-isdn
language = en
echocancel = speex
ports = +0001+0002
extension = +
number = +
cfnotify = no
cfu = no
cfnr = no
cfb = no
aocd = no
colp = no
redir = no
notify = yes
callerid = +
providetones = yes
;crypto = no
master = yes
--
asterisk*CLI> Srx reload
Feb 5 20:55:23 WARNING[4900]: chan_sirrix.c:6961 __reload: Reload of
Sirrix driver configuration will take about 2 seconds!
Feb 5 20:55:23 NOTICE[4900]: layer1_user.c:474 l1u_master_set: Setting
INTERNAL as master
Feb 5 20:55:23 NOTICE[4155]: layer1_user.c:607 l1u_run_read:
L1M1_SHUTDOWN|CONFIRM for port
Feb 5 20:55:23 NOTICE[4155]: layer1_user.c:607 l1u_run_read:
L1M1_SHUTDOWN|CONFIRM for port 0001
Feb 5 20:55:23 NOTICE[4155]: layer1_user.c:607 l1u_run_read:
L1M1_SHUTDOWN|CONFIRM for port 0002
== Parsing '/etc/asterisk/sirrix.conf': Found
Feb 5 20:55:25 NOTICE[4900]: chan_sirrix.c:6974 __reload: Sirrix
driver configuration reloaded!
Feb 5 20:55:26 NOTICE[4155]: layer1_user.c:211 l1u_from_kernel:
L1M1_STARTUP|CONFIRM for port
Feb 5 20:55:26 NOTICE[4155]: layer1_user.c:211 l1u_from_kernel:
L1M1_STARTUP|CONFIRM for port 0001
Feb 5 20:55:26 NOTICE[4155]: layer1_user.c:211 l1u_from_kernel:
L1M1_STARTUP|CONFIRM for port 0002
asterisk*CLI> Srx show chans
Port Chan InUse
0x 0x01 0
0x 0x02 0
0x0001 0x01 0
0x0001 0x02 0
0x0002 0x01 0
0x0002 0x02 0
asterisk*CLI> Srx show layers
MASTER: internal
l1=091dc668: port=0x, type=BA, mode=TE, ptp=0, ma=1, act=0:
l2h=b510ff20
l2=b5100fa8: tei= -1, st='ST_L2_1_TEI_UNASSIGNED':
l3h=b5111368: cr_out={ }, cr_in={ }
l1=091e0868: port=0x0001, type=BA, mode=TE, ptp=0, ma=1, act=0:
l2h=b51165e8
l2=b51168f0: tei= -1, st='ST_L2_1_TEI_UNASSIGNED':
l3h=b5103e00: cr_out={ }, cr_in={ }
l1=091e0270: port=0x0002, type=BA, mode=TE, ptp=0, ma=1, act=0:
l2h=b5109098
l2=b510a018: tei= -1, st='ST_L2_1_TEI_UNASSIGNED':
l3h=b510a828: cr_out={ }, cr_in={ }
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Re: [Asterisk-Users] IP PAX gateway to PSTN
Sam, I am still unsure to understand your question :-/ How much your telco is going to charge you for the PSTN calls depends on your arrangement with the telco... Usually, with proper volume interconnects (say you order a PRI line), these calls are charged per second. Do I really need PRI T1 line when I initially setup VoIP network? It largely depends what you are trying to achieve. But if you want to become a VoIP carrier for your area, yes. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] re: questions about sip requests to asterisk 1.2
hi all,
I keep asking the question and getting no replies, so i'll keep asking :-)
In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically
rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to [EMAIL PROTECTED], it will make it
[EMAIL PROTECTED]:5070, and asterisk will match it to 151 in the dialplan.
In asterisk 1.2 asterisk completely ignores the request (even at most verbose level) and an ngrep shows a "not found" returned to SER.
anyone have any idea why this is happening, bug/feature, or how to get it to work the way it did in 1.09? I want to upgrade but I don't want to lose this functionality.
thanks for any help,
yair
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Re: [Asterisk-Users] User web portal for Asterisk
On 2/4/06, Technical Support <[EMAIL PROTECTED]> wrote: > Is there a web portal available for users to: destar configures you asterisk, but also has a user-login to change some user-settings. http://destar.berlios.de/ cheers ___ --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] hardware and network requirements
On 2/3/06, John Jensen <[EMAIL PROTECTED]> wrote: > > Can a normal server with > > Pentium 4 3.6 Ghz CPU > Most likely. It'll do 40-50 concurrent 711 to 729 transcodings. Hm, interesting. In the case that you do PRI (or BRI) to G729. How do you calculate this number (40-50) ? Or do you write this number down because of your own experience? I assume when using PRI -> G.711, that machine could handle 'much' more? Cheers, Kristof. ___ --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] Search for Links for "Communicating PC to PC in the same lan through Asterisk "
John Joseph wrote: I am trying to do some simple experiment with Asterisk . my intention is to communicated two PC in start your experiment with [EMAIL PROTECTED] (http://asteriskathome.sourceforge.net/), it's very good to start with. cheers ___ --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] IP PAX gateway to PSTN
Jean-Michel Hiver wrote: Thanks for the answer. Is this PSTN gateway is something for a VoIP company to setup in order to connect their VoIP calls to the Telco's PSTN then to the end phone? I don't think Australia treat this as illegal. But I m not sure how much the Telco will charge from IP PAX (or PSTN) gateway to end phone. Assuming that there are 1000 VoIP calls thru Telco's PSTN to end phones, how will these calls get calculated? is the charge will be per-call basis? Sam, I am still unsure to understand your question :-/ How much your telco is going to charge you for the PSTN calls depends on your arrangement with the telco... Usually, with proper volume interconnects (say you order a PRI line), these calls are charged per second. Do I really need PRI T1 line when I initially setup VoIP network? Sam. As your volume increases you will usually be in a position to negociate better rates. Cheers, Jean-Michel. ___ --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] IP PAX gateway to PSTN
[EMAIL PROTECTED] wrote: Thanks for the answer. Is this PSTN gateway is something for a VoIP company to setup in order to connect their VoIP calls to the Telco's PSTN then to the end phone? I don't think Australia treat this as illegal. But I m not sure how much the Telco will charge from IP PAX (or PSTN) gateway to end phone. Assuming that there are 1000 VoIP calls thru Telco's PSTN to end phones, how will these calls get calculated? is the charge will be per-call basis? There are lots of Asterisk users here in Australia...and it's not illegal. You will probably have to discuss charging with your Telco. PaulH _ Thanks for the answers. I really appreciate that. It may be better for me to talk to local Telco for further price negotiation. Thanks Sam ___ --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] IP PAX gateway to PSTN
> Thanks for the answer. Is this PSTN gateway is something for a VoIP > company to setup in order to connect their VoIP calls to the Telco's > PSTN then to the end phone? I don't think Australia treat this as > illegal. But I m not sure how much the Telco will charge from IP PAX (or > PSTN) gateway to end phone. Assuming that there are 1000 VoIP calls thru > Telco's PSTN to end phones, how will these calls get calculated? is the > charge will be per-call basis? There are lots of Asterisk users here in Australia...and it's not illegal. You will probably have to discuss charging with your Telco. PaulH ___ --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: delaying "answer" for a number of rings or an amount
> > Here's a step-by-step of what happens below: > > 1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds. > > So you don't want Asterisk to wait and see if the POTS line is picked up > before ringing the SIP phones? Interesting. If it's anything like my setup, Asterisk handles ALL calls, whether from sip, iax, or zap. So when the zap line rings, asterisk will ring your internal sip phone(s), and if the call isn't picked up after so many seconds, it'll stop ringing the internal lines and go straight to voicemail. No phones are connected directly to the POTS line, just asterisk. The only downside to this approach, is the caller will hear about two rings before you beging to hear anything (takes asterisk that long to see the call, check for callerid, then start ringing your internal lines). My solution is to have a quick greeting played to the caller, then they hear ringing again when the internal lines ring. Also gives me a chance to force callers to press "1" if I don't recognize their callerid, stops telemarketers dead in their tracks (those auto-dialing machines that ring you and either hang up after you pick up, or tell you to stay on the line for an important message, will not know to dial 1 first and will be hung up on). > > 2 - After 30 seconds if the line is still ringing (nobody picked up POTS > > phone or SIP phones) * answers the line and sends to Voicemail. Asterisk > > never picks up the call until the 30 seconds are up. > > What seems to be happening here is that even if somebody picks up the > POTS line within a few seconds, after the 30 seconds (Wait() in my case, > but I'd imagine the same will happen after ringing the SIP lines for > 30s) is up Asterisk is also on the POTS line (with the callee who picked > up the POTS phone) doing the voicemail intro and recording the > conversation. Again, give everyone in your home/office a phone connected to asterisk (whether it's a sip/iax phone, or a regular phone connected to an ATA, or what have you). Any call that comes in will go through asterisk. Then you won't have to worry about having it detect if a POTS line was picked up directly, if you have it pass the call to an internal phone, it'll know if that phone picked up or not, and will know whether to pass it to voicemail or not. Joseph Tanner > > [from-pots] > > exten => s,1,Dial(SIP/brian&SIP/joe,30) > > exten => s,2,Voicemail(u2001) > > exten => s,3,Hangup > > I will try this exactly and see if it works any better. > > b. > > -- > My other computer is your Microsoft Windows server. > > Brian J. Murrell > > > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.2 (GNU/Linux) > > iD8DBQBD45ffl3EQlGLyuXARAobbAJoCaGeIV/gzNTyfw1h6xt+EYCdHPwCeIwfZ > J3CaPbHa1j3wxqJw/aK9+NY= > =ttIm > -END PGP SIGNATURE- > > > ___ > --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
[Asterisk-Users] Codec Selection
Hi All, I have one Carrier which is supporting only G.723.1, how i can put in my extentions.conf to send calls to this GW using G.723.1, because for Clients i can specify the codec from sip.conf but i am little confiuse how i can give specific codec for carriers. your ideas will be appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] List Broken Again?
Sorry, loaner laptop that doesnt have a rule to file asterisk users posts to a different folder. My mistake. -Original Message- From: Steve Totaro Sent: Sun 2/5/2006 5:44 AM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] List Broken Again? last message I have received was on1/27/06. <>___ --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] List Broken Again?
last message I have received was on1/27/06. ___ --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] IP PAX gateway to PSTN
Thanks for the answer. Is this PSTN gateway is something for a VoIP company to setup in order to connect their VoIP calls to the Telco's PSTN then to the end phone? I don't think Australia treat this as illegal. But I m not sure how much the Telco will charge from IP PAX (or PSTN) gateway to end phone. Assuming that there are 1000 VoIP calls thru Telco's PSTN to end phones, how will these calls get calculated? is the charge will be per-call basis? Sam, I am still unsure to understand your question :-/ How much your telco is going to charge you for the PSTN calls depends on your arrangement with the telco... Usually, with proper volume interconnects (say you order a PRI line), these calls are charged per second. As your volume increases you will usually be in a position to negociate better rates. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] IP PAX gateway to PSTN
Jean-Michel Hiver wrote: Sam a écrit : Hi, If I setup an IP PAX gateway to handle VoIP calls to a traditional phone line, I am wondering how each VoIP call to the PSTN connection get charged by a local Telecom. I am not really sure to understand the question. But assuming you are having: (remote phone) -> internet -> PSTN gateway -> end phone The connection charge is going to be PSTN gateway -> end phone. However note that in certain VoIP-backwards countries this scheme is illegal, and the telco might ask you to pay the international call termination charge if they find out you're doing this. Thanks for the answer. Is this PSTN gateway is something for a VoIP company to setup in order to connect their VoIP calls to the Telco's PSTN then to the end phone? I don't think Australia treat this as illegal. But I m not sure how much the Telco will charge from IP PAX (or PSTN) gateway to end phone. Assuming that there are 1000 VoIP calls thru Telco's PSTN to end phones, how will these calls get calculated? is the charge will be per-call basis? thanks Sam Cheers, Jean-Michel. ___ --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] ddi???
On 4 Feb 2006, at 23:33, Chris Bagnall wrote:You need to get BT to agree and allocate or port the numbers.You need to agree how many digits BT will pass on to you (probably 1925838395 but possibly just the last 2) I don't know the number of digits that BT pass through on a PRI, but on aset of BRIs with a range of DDIs, they're passing the last 6 digits (sogiven the OP's range, you'd want to match on 838381 etc.)I concur with Tim's suggestion of trying to get the internal extensionsrelated to the DDIs - it'll simplify your dialplan substantially.Out of curiosity, why do you want to go to BT for the number range? 8channels through BT will cost a small fortune, and you could run 8concurrent calls over a standard ADSL connection in the UK with appropriatecodec selections. There are at least 3 or 4 companies in the UK that'lloffer you a consecutive number range for a UK area code.Only a small fortune though :-) my 8 NTL lines are £13/month (each)BT do a similar deal. (what is a business grade ADSL line now? £50/month?)Where a VOIP supplier you might save big-time would be on the callcosts.If you sign up with a VOIP provider for business purposes,make _absolutely sure_ you understand the risks.Check the SLA and compare to BT/NTL'sCheck your ISP's SLA.Make sure you own the numbers and can port them offto another provider (or a traditional telco).I looked at these factors and decided that VOIP was too risky forour main number, but fine for 'extras' and low cost international.This was based on an experience we had 18 months ago,BT had a major fire in the local exchange trunk in Manchester.They had the phones working or redirected to mobiles withinhours. (NTL just kept working as they were on a fibre that didn'tpass through that duct). Our ISP was unable to offer any sortof service for the best part of 10 days, and we were paying£6k/yr for the leased line. If our phones had been over thatwe would have been out of business for 10 days. You'd also avoid a substantial chunk of potential echo issues. The asteriskdeployments we've done where the client has had calls delivered via IAX froma provider have all been *much* easier and taken far less time than when wehave to fight with ISDN lines, or worse, analogue lines.I'm pretty happy with my Digium PRI card, and it isn't even one withecho canceling hardware. I haven't touched analog or BRI, asyet.Regards,Chris-- C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.westhawk.co.uk/ ___ --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] IP PAX gateway to PSTN
Sam a écrit : Hi, If I setup an IP PAX gateway to handle VoIP calls to a traditional phone line, I am wondering how each VoIP call to the PSTN connection get charged by a local Telecom. I am not really sure to understand the question. But assuming you are having: (remote phone) -> internet -> PSTN gateway -> end phone The connection charge is going to be PSTN gateway -> end phone. However note that in certain VoIP-backwards countries this scheme is illegal, and the telco might ask you to pay the international call termination charge if they find out you're doing this. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] Search for Links for "Communicating PC to PC in the same lan through Asterisk "
On Sun, Feb 05, 2006 at 07:51:33AM +, John Joseph wrote: > Hi > I am trying to do some simple experiment with > Asterisk . my intention is to communicated two PC in > my lan to voice -communicate with each other with out > extra hardware > I searched the FAQ and wiki for any links > for this , so far I have not found one , It would be > much help , if I get a link on communicating PC > to PC in the same lan through Asterisk Here are some relevant keywords: you basically need to install softphones (software VoIP "phones") on the computers that use either SIP or IAX. then set up in Asterisk extensions for both of them (SIP or IAX, depends on the type of the softphone). -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] IP PAX gateway to PSTN
Hi, If I setup an IP PAX gateway to handle VoIP calls to a traditional phone line, I am wondering how each VoIP call to the PSTN connection get charged by a local Telecom. Thanks Sam ___ --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
答复: [Asterisk-Users] Search for Links for "Communicating PC to PC inthe same l an through Asterisk "
You can use SIP soft phone to do pc to pc voice communication witch asterisk. You need only to define two users at sip.conf such as 5000 and 5001 Then make a simple extension define in extension.conf exten => 5000,1,Dial(SIP/5000,20) exten => 5001,1,Dial(SIP/5001,20) -邮件原件- 发件人: John Joseph [mailto:[EMAIL PROTECTED] 发送时间: Sunday, February 05, 2006 3:52 PM 收件人: asterisk-users@lists.digium.com 主题: [Asterisk-Users] Search for Links for "Communicating PC to PC inthe same lan through Asterisk " Hi I am trying to do some simple experiment with Asterisk . my intention is to communicated two PC in my lan to voice -communicate with each other with out extra hardware I searched the FAQ and wiki for any links for this , so far I have not found one , It would be much help , if I get a link on “ communicating PC to PC in the same lan through Asterisk “ Thanks Joseph John ___ Yahoo! Photos �C NEW, now offering a quality print service from just 8p a photo http://uk.photos.yahoo.com ___ --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
