Re: [Asterisk-Users] IVR
It is very feasable. Using AGI you can pretty much connect to db's using perl, c, etc... what kind of database are you thinking about using? Omar Matthew Murray wrote: Hi, I am looking into Asterisk as a small startup call center platform. Is there any detailed information on what IVR features are available? Such as Database capabilities and if the system can respond during a call? Sorry if this has been asked before, I searched through the archives before posting. Thanks. Matt _ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Hi everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that when I use g723 I can call MGCP but I can't call PSTN (call goes off when I pick the phone up). What can I do? 2)Whatis [EMAIL PROTECTED] ? For what is used? 3)Can I transfer calls? I guess that if I do transfer = yes in the general section of sip.conf, it should work... but it doesn't!! 4)And finally, the caller ID. I have done usecallerid=yes in the general section of sip.conf and the I put callerid="SIP" in the [sip] section (the one that I have created for my devide). But it doesn't work either! Any ideas? My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no [sip] type=friend callerid="sip" username=sip host=188.208.12.37 accountcode=sip Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] some sip questions
I write the email again, cause the first one I have had problems while sending it. Here is the email again: Hi everybody, I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that when I use g723 I can call MGCP but I can't call PSTN (call goes off when I pick the phone up). What can I do? 2)What is [EMAIL PROTECTED] ? For what is used? 3)Can I transfer calls? I guess that if I do transfer = yes in the general section of sip.conf, it should work... but it doesn't!! 4)And finally, the caller ID. I have done usecallerid=yes in the general section of sip.conf and the I put callerid="SIP" in the [sip] section (the one that I have created for my devide). But it doesn't work either! Any ideas? My sip.conf:[general]port = 5060bindaddr = 0.0.0.0context = defaulttransfer = yesthreewaycalling = yesusecallerid = yeshidecallerid = no [sip]type=friendcallerid="sip" username=siphost=188.208.12.37accountcode=sip Thanks you all!!! Michelle Tu cuenta de correo gratuita Mixmail Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] some sip questions AGAIN
I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that when I use g723 I can call MGCP but I can't call PSTN (call goes off when I pick the phone up). What can I do? 2)What is [EMAIL PROTECTED] ? For what is used? 3)Can I transfer calls? I guess that if I do transfer = yes in the general section of sip.conf, it should work... but it doesn't!! 4)And finally, the caller ID. I have done usecallerid=yes in the general section of sip.conf and the I put callerid=SIP in the [sip] section (the one that I have created for my devide). But it doesn't work either! Any ideas? My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no [sip] type=friend callerid=sip username=sip host=188.208.12.37 accountcode=sip Thanks you all!!! Michelle - Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Opportunistic VoIP
This is slightly off-topic I suppose, but: At 20:37 10-6-2003 -0700, you wrote: You should investigate TRIP (RFC 3129): http://www.zvon.org/tmRFC/RFC3219/Output/ Find BSD-licensed proof-of-concept code at http://www.vovida.org/downloads/trip/trip-1.0.0.tar.gz If someone could incorporate this into Asterisk and extend the functionality, that would be pretty nice. The basic ENUM support in Asterisk already can handle specific number paths, but I think TRIP or something like TRIP would be best for handling situations where larger groups of numbers need to be advertised into a routing table behind a particular Asterisk server. Think BGP for phone numbers. I'm sorry, but I see no real benefits to TRIP over ENUM. Large amounts of data in DNS databases have not been a real problem yet, provided the tree is delegated properly (as ENUM does), and works quite effectively due to caching. TRIP only makes it harder for widespread use to deal with such things as number portability (can't ever do that with IP, remember). As far as I can tell from the TRIP docs this looks a lot like some big telco tries to make it more difficult for customers to move to another telco and still use their old number... Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P Setup
Can anyone give us a clue on setting up a E100P we just get Busy tone all the time. The LED on the back of the card shows green which I assume is good. Regards Mark McKibbin DCS Internet 64 Queen St Warragul Victoria3820 Australia www.dcsi.net.au [EMAIL PROTECTED] Ph. 1300 665575 Fx. 1300 556595 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NewbieQ: SOHO setup with x100p
On Tue, 10 Jun 2003, Tielman Koekemoer wrote: welcome to * Good to be here. c) get a T400P + channel bank (expensive, but it does give you 24 ports) I'm also considering a PRI from our local Telco (thanks to Mr Davies)connected to an E100P but am waiting for a quote from said Telco to compare costs with your idea (T[1,4]00P + channel bank). If any, what are the advantages of either? Off the cuff please, I don't expect lengthy answers. Would I need a channel bank with the E100P? I'm asking as I need to keep costs low. theoretically, * can route fax calls for you, see exten = fax practically, you're welcome to contribute your experiences to the list :) Thanks! -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems PK #48, St 32, Sector F-6/1, Islamabad, Pakistan 44000 | US: +1(800)460-1446 VOX: +92(51)282-0628 | FAX: +92(51)282-0621 | GSM: +92(300)850-8070 This mail is confidential intended solely for the use of the addressee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Content and Virus scanned by Inflex and Mcafee _ Content and Virus scanned by Inflex and Mcafee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie : i try and test to use asterisk
I try to use X-lite with asterisk on intranet In sip.conf i have [general] port = 5060 bindaddr = 0.0.0.0 context = default [roseau] type=friend host=dynamic dtmfmode=inband context=sip [bambou] type=friend host=dynamic dtmfmode=inband context=sip and in extensions.conf [sip] exten = 1000,1,Dial,SIP/roseau exten = 2000,1,Dial,SIP/bambou i use X-Lite on windows in setup ; Display name : roseau user name : 1000 authorization user : same as user name Password : Domain/Realme : 192.168.0.2 SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty i obtain in /var/log/messages when i try to call [handle_request]: Registration from 'roseau 'sip:[EMAIL PROTECTED]' failed for '192.168.0.4' Is anybody help me to start please regards (and very sorry for my english) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to receive call on iaxclient
Hello, I have successfully tested the new IAXCLIENT release (even with GUI) to initiate calls. I wonder now how I could receive call on this client (using dynamic IP address) as I didn't see any kind of registration. Thanks and regards. Francois. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All extensions busy
Hi Firstly could I thnk everyone who has helped me so far, I just have a couple of queries I have not had chance to debug this much yet but When using the tdm40p all extesions busy themselves out, and * cannot rint the extensions for incoming calls is this because I don't have a hangup statement at the end of the incoming context? if not has anyone any idea? does anyone have a quick and dirty IAX confiuration sample Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie : i try and test to use asterisk
i use X-Lite on windows in setup ; Display name : roseau user name : 1000 authorization user : same as user name Password : Domain/Realme : 192.168.0.2 SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty You're using a username that is different from [username] in sip.conf. You can configure X-lite as username: roseau for example, or to add in sip.conf, for each phone, a username=x field. [roseau] type=friend host=dynamic dtmfmode=inband context=sip username=1000 -- Stefano ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie : i try and test to use asterisk
Hi, You need to change your settings in X-lite: Display name : roseau user name : 1000 --- this is wrong! authorization user : same as user name Password : Domain/Realme : 192.168.0.2 SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty to: user name : roseau (That should match the definition in sip.conf so on the other Pc you would use bambou) authorization user : same as user name Password : Domain/Realme : ip of the pc xlite is running on SIP Proxy : ip of asterisk box:5060 HTH Andy *** REPLY SEPARATOR *** On 11/06/2003 at 11:11 Hervé THIBAUD wrote: I try to use X-lite with asterisk on intranet In sip.conf i have [general] port = 5060 bindaddr = 0.0.0.0 context = default [roseau] type=friend host=dynamic dtmfmode=inband context=sip [bambou] type=friend host=dynamic dtmfmode=inband context=sip and in extensions.conf [sip] exten = 1000,1,Dial,SIP/roseau exten = 2000,1,Dial,SIP/bambou i use X-Lite on windows in setup ; Display name : roseau user name : 1000 authorization user : same as user name Password : Domain/Realme : 192.168.0.2 SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty i obtain in /var/log/messages when i try to call [handle_request]: Registration from 'roseau 'sip:[EMAIL PROTECTED]' failed for '192.168.0.4' Is anybody help me to start please regards (and very sorry for my english) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Underwater in 10 - 20 seconds
Title: Underwater in 10 - 20 seconds I'm running a X100P connected to a POTS line and a TDMP400P w/ two FXS daughter cards. Both calling out from one of the FXS phones (internally) or calling my home number (externally) the FXO card starts to freak out. By freak out I mean I can still hear but it sounds like you are underwater, there is an annoying hiss or buzz on the line as well. If I hang up and pick up another house phone the hiss and buzz is still on the other house lines. The only thing that fixes it is rebooting the asterisk machine / or unplugging the FXO card from the wall. Any ideas on what is causing this to happen or what might be done to fix it? Calling from one FXS phone to the other has worked just great but I would like to be able to talk to someone other than myself! -Matt
[Asterisk-Users] Dialing out through a Hardware PBX
hello All, our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9 to take an outside call through the hardware pbx, our fxo interface is also connected to one of the extensions of it. we can make calls to internal hardware pbx extensions by dialing through the fxo interface using Dial function, for ex. Dial(Zap/g3/599|20|t) but we also want make calls to outside by first dialing 9, and then dialing the number. is there any possiblity that asterisk can make calls like that, ie, first dialing 9, and then wait for the dial tone and then dialing the number? how do i pause between 9 and the telephone number, will comma ( , )do the job? for ex. will Dial(Zap/g3/9,001338|20|t) will work? or else pls let me know a way to do that. Thanx inadvance, Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing out through a Hardware PBX
On Wed, 11 Jun 2003 [EMAIL PROTECTED] wrote: is there any possiblity that asterisk can make calls like that, ie, first dialing 9, and then wait for the dial tone and then dialing the number? how do i pause between 9 and the telephone number, will comma ( , ) do the job? for ex. will Dial(Zap/g3/9,001338|20|t) will work? or else pls let me know a way to Dial(Zap/g3/9w0777blah|20|t) etc, iirc a , will terminate that field according to the old dialplan syntax -wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phone behind NAT
Hi all, I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a we do not rely. I tried to forward the SIP Port (5060) UDP via UPnP to the internal Snom100 IPadress and a port range forwarding of 16384 - 32768 (UDP) for the RTP traffic. Additionally I tried to change host = dynamic to host = myserver.dyndns.org to ensure the SIP traffic is going to my Linksys ADSL router and be forwarded to the internal SIP 100 phone. But all my effort did not have success. Any suggestions ?? regards Olaf extensions.conf - [sip] ; will be replaced by a macro exten = 2123,1,Dial(SIP/snom200,30,tr) exten = 2123,2,Playback(new/nbdy-avail-to-take-call) exten = 2123,3,Voicemail(u2123) exten = 2123,4,Hangup exten = 2123,102,Voicemail(b2123) exten = 2123,103,Hangup exten = 2124,1,Dial(SIP/snom100,30,tr) exten = 2124,2,Playback(new/carried-away-by-monkeys) exten = 2124,3,Voicemail(u2124) exten = 2124,4,Hangup exten = 2124,102,Playback(new/lots-o-monkeys) exten = 2124,103,Voicemail(b2124) exten = 2124,104,Hangup sip.conf -- [snom200] type=friend secret=snom200 host=ip-adress-of-snom200 dtmfmode=rfc2833 mailbox=2123 context=sip callerid=snom200 2123 [snom100] type=friend secret=snom100 host=dynamic nat=1 dtmfmode=rfc2833 mailbox=2124 context=sip callerid=snom100 2124 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Testing two E400P with E1 cross-cable
Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1Server#2 caller callee +---+ +---+ | Span1 |--E1 crossover cable--| Span1 | | Span2 |--E1 crossover cable--| Span2 | | Span3 |--E1 crossover cable--| Span3 | | Span4 |--E1 crossover cable--| Span4 | +---+ +---+ The basic configuration seems ok, since zttool shows the links are OK. I'm using this UTP cat5 cross-cable (not shielded): pin1 -- pin4 pin2 -- pin5 pin3 -- pin6 pin4 -- pin1 pin5 -- pin2 pin6 -- pin3 pin5 -- pin8 pin8 -- pin7 I'm not really sure this is correct, since I've found only how to connect pins 1,2,4 and 5. The other pins are connected as I supposed it should be. This is my zapata.conf: Server#1 zapata.conf (Server#2 has the same zapata.conf but pri_net is pri_cpe) -- [channels] context=inicio switchtype=euroisdn signalling=pri_net rxwink=300; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 signalling=pri_net channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 -- To generate calls, I've done a small C proggie that generates files in /var/spool/asterisk/outgoing/ like this one (filename and callerid are different for each call): -- Channel: Zap/g1 Context: default Extension: s Priority: 1 Callerid: 55512345 -- The Dialplans are simple...caller machine just plays a 3 minutes gsm and loops, and the callee machine dilaplan launches an AGI that plays some gsm, records 20 secs of the call and hangups the call. Server#1 (caller) extensions.conf -- [general] static=yes writeprotect=no [inicio] exten = s,1,PlayBack(laxana) exten = s,2,Goto(s,1) exten = t,1,hangup exten = i,1,hangup exten = o,1,hangup exten = h,1,hangup -- Server#2 (callee) extensions.conf -- [general] static=yes writeprotect=no [inicio] exten = s,1,Answer exten = s,2,Agi,600agi exten = s,3,hangup exten = t,1,hangup exten = i,1,hangup exten = o,1,hangup exten = h,1,hangup -- Everything works pretty well with just 1 call, but my goal is to generate a much high number (a queue of +50.000 calls), but I've run into some problems... If I try to generate 120 simultaneous calls (top of lines available with my 4 E1), I get a lot of errors in the logs. I've tried to sleep(1) the begining of each call, and now I can only generate a maximum of 60~80 simultaneous calls. Could it be due to a cross-over cable problem or maybe the server can't deal with it? Btw, asterisk gets 100% of all available CPU (user CPU) for 10~20 seconds and then it keeps about 80% CPU usage Thas's just a portion of the asterisk log: -- Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: !! Got reject for frame 67, retransmitting frame 67 now, updating n_r! Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: !! Got reject for frame 67, retransmitting frame 68 now, updating n_r! Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: !! Got reject for frame 67, retransmitting frame 69 now, updating n_r! (...) Jun 11 13:12:16 WARNING[90124]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: Short write: -1/16 (Unknown error 500) Jun 11 13:12:16 WARNING[90124]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: Short write: -1/20 (Unknown error 500) Jun 11 13:12:16 WARNING[90124]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: Read on 138 failed: Unknown error 500 (...) Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: !! Got reject for frame 67, retransmitting frame 70 now, updating n_r! Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: Short write: -1/20 (Unknown error 500) Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: !! Got reject for frame 67, retransmitting frame 71 now, updating n_r! Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341 (zt_pri_error): PRI: Short write: -1/16 (Unknown error 500) Jun 11 13:12:16 WARNING[81931]: File chan_zap.c, Line 5341
[Asterisk-Users] Configuring zhone zplex to 24 fxs ports
Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -BEGIN GEEK CODE BLOCK- Version: 3.12 GS d? s:- a- C+++ UL P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e h! r++ y --END GEEK CODE BLOCK-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring zhone zplex to 24 fxs ports
Sure, contact the sales department at Digium. Jeremy McNamara Brancaleoni Matteo wrote: Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing two E400P with E1 cross-cable
Do you see in /proc/interrupts that tor2 receives IRQs on both CPUs ? Martin On Wed, 11 Jun 2003, Carlos Carús wrote: Martin Pycko escribió: Did you recompile zaptel for -D__SMP__ ? Check the zaptel/Makefile Martin Yes, I did :-( -- Carlos Carús Ingeniero de Sistemas [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Alisys Software Alisys Software, S.L. Edificio Lexington - C/ Orense, 85 28020 MADRID Tfno.: 985175935 - 915678474 Fax: 915714244 web: http://www.alisys.net http://www.alisys.net/ wap: http://www.alisys.net/wap/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone behind NAT
Hi Olaf, I've just started working on a SIP and RTP proxy to handle exactly this. I'm really just in proof of concept at the moment but just one hour ago I got a completely successful connection out over NAT in which both endpoints thought they were talking to the proxy. I should have the code posted in the next few days. So far it's only tested under Linux but it should work on Windows without too many problems. I'll post more info in the next few days but feel free to email me directly if you are inerested or haven't heard anything from me. Regards, Andrew Radke Olaf Menzel wrote: Hi all, I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a we do not rely. I tried to forward the SIP Port (5060) UDP via UPnP to the internal Snom100 IPadress and a port range forwarding of 16384 - 32768 (UDP) for the RTP traffic. Additionally I tried to change host = dynamic to host = myserver.dyndns.org to ensure the SIP traffic is going to my Linksys ADSL router and be forwarded to the internal SIP 100 phone. But all my effort did not have success. Any suggestions ?? regards Olaf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing two E400P with E1 cross-cable
Martin Pycko escribió: Do you see in /proc/interrupts that tor2 receives IRQs on both CPUs ? Martin On Wed, 11 Jun 2003, Carlos Carús wrote: Yes, it does too: $cat /proc/interrupts CPU0 CPU1 0: 410297 0 local-APIC-edge timer 1: 2 2IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 0 1IO-APIC-edge rtc 11: 0 0 IO-APIC-level usb-ohci 12: 11 9IO-APIC-edge PS/2 Mouse 14: 0 2IO-APIC-edge ide0 20: 71136 55323 IO-APIC-level tor2 25: 8 5 IO-APIC-level eth1 27:383466 IO-APIC-level eth0 28: 2462 2406 IO-APIC-level aic7xxx NMI: 0 0 LOC: 410040 410039 ERR: 0 MIS: 0 I've cheched for number of available inodes too, memory usage, etc.. I think I have a zaptel misconfig, but I can't see anything wrong... Or maybe I've made a wrong pinout in the E1 cross cable... I'm using a standar UTP cat5 cable, should I need to use shielded cable? Does anyone know the specifications of a cross cable for these tests? Greeting from a totally lost asterisk user! -- Carlos Carús Ingeniero de Sistemas [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Alisys Software Alisys Software, S.L. Edificio Lexington - C/ Orense, 85 28020 MADRID Tfno.: 985175935 - 915678474 Fax: 915714244 web: http://www.alisys.net http://www.alisys.net/ wap: http://www.alisys.net/wap/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring zhone zplex to 24 fxs ports
On Wednesday 11 June 2003 09:04 am, Brancaleoni Matteo wrote: I've already bought it (2 of them) ;) So was wondering if anyone has a hint or a restore file to put it into all fxs mode... There are two different models of the Zhone Zplex 10B. One is a combination 8/16 and the other is 24 FXS. AFAIK, there is no way to convert one to the other. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:Some SIP questions AGAIN
Hi Edwin I have my mgcp.conf almost the same as yours, except from nat=1 , why do you put it? Anyway, DL102s now works more or less acceptably so now I'm having a battle with sip.conf Thank you for your help Michelle - Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i make best use of Macro?
On Wednesday 11 June 2003 10:43 am, Christopher Arnold wrote: Hi, im trying to setup a chat system. And i belive the best way is using an macro. But a couple of questions regarding using macros pops up. a) Is there state building up if my macro calls itself recusivly? A macro is NOT a function. It simply is a shortcut to doing a longer series of commands. A macro cannot itself be recursive. b) How do i return from a macro? One does not return from a macro. Indeed, once the macro is expanded, it is no longer part of the process. Or should i use a Goto(newcontext,s,1)? I think you're confused as to what a macro can and cannot do. Until you understand that distinction, you're going to have trouble understanding how a macro works within the system. In any case, I'd recommend that you forget about using macros and instead use AGI. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re:Some SIP questions AGAIN
Nat=1 is so that mgcp functions properly behind a NAT gateway. What kind of problems are you having with your SIP? What type of SIP phone do you have? Can you elaborate a little more or even post you SIP.conf? Here's what ours looks like so you can do a comparison: Sip.conf --- ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sipstart ; Default for incoming calls tos = lowdelay [sip_phone] type=friend username=sip_phone secret=sip_phone host=dynamic nat=1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of michelle matis litio Sent: Wednesday, June 11, 2003 12:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re:Some SIP questions AGAIN Hi Edwin I have my mgcp.conf almost the same as yours, except from nat=1 , why do you put it? Anyway, DL102s now works more or less acceptably so now I'm having a battle with sip.conf Thank you for your help Michelle - Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ¡Gratis! http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] lost variables
Hi, Seems that my local variable content get lost when I call an AGI program. Is this the correct functionality? Thanks, Paulo H. Mannheimer
RE: [Asterisk-Users] Bandwidth measurement tool: bmtools
Looks like they changed their site to http://s-tech.elsat.net.pl/bmtools/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Bourg Sent: Wednesday, June 11, 2003 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Bandwidth measurement tool: bmtools I can't resolve this host from anywhere. Is there a mirror somewhere? Thanks, Steve Bourg On Sat, 7 Jun 2003, John Todd wrote: This is not specifically on-topic for Asterisk, but I have found on many occasions while working with Asterisk that it would have been very handy to be able to measure, with some precision, the bandwidth being used by a particular host, port, or combination of the two. So, I went searching for various tools, none of which were what I wanted. They either were too clever, or too limited in their abilities. However, someone forwarded the link to this tool to me about an hour ago, and I've been thrilled that it does _exactly_ what I want. I can use a BPF-style filter to monitor exactly what I'd like to watch, and it hands back results to me in real time down to a one-second interval. Sometimes, a small program can make me very happy, and I suppose after a morning full of various system problems I'm overly happy have something that works and does just what I want it to. This is useful for checking to see how much bandwidth a codec _really_ uses, or seeing what your total usage is between two IAX hosts, or pretty much anything that requires live examination of ethernet segment traffic. http://s-tech.linux-pl.com/bmtools/ [EMAIL PROTECTED] bmtools-0.71]# ./rate -r 1 -f 'host 10.0.1.3 and not port ssh' - Currently 263.05 Bps/3.01 pps, Average: 263.05 Bps/3.01 pps - Currently 2706.00 Bps/17.00 pps, Average: 1486.97 Bps/10.02 pps - Currently 588.00 Bps/6.00 pps, Average: 1186.92 Bps/8.68 pps - Currently 440.00 Bps/4.00 pps, Average: 1000.00 Bps/7.51 pps - Currently 440.00 Bps/4.00 pps, Average: 887.91 Bps/6.81 pps - Currently 2080.00 Bps/16.00 pps, Average: 1086.72 Bps/8.34 pps - Currently 1282.00 Bps/9.00 pps, Average: 1114.64 Bps/8.43 pps - Currently 10385.00 Bps/20.00 pps, Average: 2274.01 Bps/9.88 pps ^C JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing two E400P with E1 cross-cable
Martin Pycko escribió: It should be good enough. The problem is propably in software configuration Martin On Wed, 11 Jun 2003, [UTF-8] Carlos Carús wrote: So if cable is ok, the problem must be one of these three: 1.- Config error, as Martin points (most probably) 2.- System can't hold the load (humm..) 3.- Faulty E400P? (don't want to beleive it) Let's suppose it's a config mistake... Server#1 (caller) zaptel.conf: --- span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 loadzone = fr defaultzone = fr Server#2 (callee) zaptel.conf: --- span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 loadzone = fr defaultzone = fr These are zapata.conf: - [channels] context=inicio switchtype=euroisdn signalling=pri_cpe (pri_net in server#2) rxwink=300; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 signalling=pri_cpe (pri_net in server#2) channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 Server#1 (caller) extensions.conf: --- [general] static=yes writeprotect=no [inicio] exten = s,1,PlayBack(laxana) exten = s,2,Goto(s,1) exten = t,1,hangup exten = i,1,hangup exten = o,1,hangup exten = h,1,hangup Server#2 (callee) extensions.conf - [general] static=yes writeprotect=no [inicio] exten = s,1,Answer exten = s,2,Agi,600agi exten = s,3,hangup exten = t,1,hangup exten = i,1,hangup exten = o,1,hangup exten = h,1,hangup Do you see something weird??? Thnaks people! -- Carlos Carús Ingeniero de Sistemas [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Alisys Software Alisys Software, S.L. Edificio Lexington - C/ Orense, 85 28020 MADRID Tfno.: 985175935 - 915678474 Fax: 915714244 web: http://www.alisys.net http://www.alisys.net/ wap: http://www.alisys.net/wap/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bandwidth measurement tool: bmtools
Title: RE: [Asterisk-Users] Bandwidth measurement tool: bmtools http://s-tech.elsat.net.pl/bmtools/ -Original Message- From: Steve Bourg [mailto:[EMAIL PROTECTED]] Sent: Wednesday, June 11, 2003 11:44 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Bandwidth measurement tool: bmtools I can't resolve this host from anywhere. Is there a mirror somewhere? Thanks, Steve Bourg On Sat, 7 Jun 2003, John Todd wrote: This is not specifically on-topic for Asterisk, but I have found on many occasions while working with Asterisk that it would have been very handy to be able to measure, with some precision, the bandwidth being used by a particular host, port, or combination of the two. So, I went searching for various tools, none of which were what I wanted. They either were too clever, or too limited in their abilities. However, someone forwarded the link to this tool to me about an hour ago, and I've been thrilled that it does _exactly_ what I want. I can use a BPF-style filter to monitor exactly what I'd like to watch, and it hands back results to me in real time down to a one-second interval. Sometimes, a small program can make me very happy, and I suppose after a morning full of various system problems I'm overly happy have something that works and does just what I want it to. This is useful for checking to see how much bandwidth a codec _really_ uses, or seeing what your total usage is between two IAX hosts, or pretty much anything that requires live examination of ethernet segment traffic. http://s-tech.linux-pl.com/bmtools/ [EMAIL PROTECTED] bmtools-0.71]# ./rate -r 1 -f 'host 10.0.1.3 and not port ssh' - Currently 263.05 Bps/3.01 pps, Average: 263.05 Bps/3.01 pps - Currently 2706.00 Bps/17.00 pps, Average: 1486.97 Bps/10.02 pps - Currently 588.00 Bps/6.00 pps, Average: 1186.92 Bps/8.68 pps - Currently 440.00 Bps/4.00 pps, Average: 1000.00 Bps/7.51 pps - Currently 440.00 Bps/4.00 pps, Average: 887.91 Bps/6.81 pps - Currently 2080.00 Bps/16.00 pps, Average: 1086.72 Bps/8.34 pps - Currently 1282.00 Bps/9.00 pps, Average: 1114.64 Bps/8.43 pps - Currently 10385.00 Bps/20.00 pps, Average: 2274.01 Bps/9.88 pps ^C JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] segmentation asterisk oh323
B. Katisi Electrical Engineer when i call the error is give *CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created. WrapH323Connection::OnReceivedSignalSetup: Received SETUP message... 0:18.185 H225 RAS:80efe50 RAS admissionRequest rejected: callerNotRegistered 0:21.171 H225 Answer:80cf028 RAS Timeout on request seqnum=3812, try #1 of 2 0:21.197 H225 RAS:80efe50 RAS admissionRequest rejected: callerNotRegistered 0:24.181 H225 Answer:80cf028 RAS Timeout on request seqnum=3812, try #2 of 2 WrapH323Connection::OnAnswerCall: User robert k [212.88.98.100] is calling us... WrapH323Connection::OnAnswerCall: Call reference: 14724 WrapH323Connection::OnAnswerCall: Call token: ip$212.88.98.100:1328/14724 WrapH323Connection::OnAnswerCall: Call source alias: robert k [212.88.98.167](24) WrapH323Connection::OnAnswerCall: Call dest alias: 668 668(7) WrapH323Connection::OnAnswerCall: Call source e164: (0) WrapH323Connection::OnAnswerCall: Call dest e164: 668(3) WrapH323Connection::OnAnswerCall: Remote Party number: WrapH323Connection::OnAnswerCall: Remote Party name: robert k [212.88.98.167] WrapH323Connection::OnAnswerCall: Remote Party address: robert [EMAIL PROTECTED]:1328 Segmentation fault linux:/etc/asterisk # Ouch ... error while writing audio data: : Broken pipe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing two E400P with E1 cross-cable
Something for you to think about, your machines should be more than powerful enough to move that much data. If the system load is high, maybe you might want to look into what file type you where playing on the line to simulate the call. If it is playing a mp3, or a GSM file there is decompression and a lot of context switching going on on top of dealing with the calls. I think you should first look into compiling your own kernel. You will sometimes see a pretty significant speed difference in just that. Then maybe look into the codecs being used. Try converting to slinear so there is no real compute load outside of the servicing of the E400P. On Wed, 2003-06-11 at 12:32, Carlos Carús wrote: Martin Pycko escribió: It should be good enough. The problem is propably in software configuration Martin On Wed, 11 Jun 2003, [UTF-8] Carlos Carús wrote: So if cable is ok, the problem must be one of these three: 1.- Config error, as Martin points (most probably) 2.- System can't hold the load (humm..) 3.- Faulty E400P? (don't want to beleive it) Let's suppose it's a config mistake... Server#1 (caller) zaptel.conf: --- span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 loadzone = fr defaultzone = fr Server#2 (callee) zaptel.conf: --- span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 loadzone = fr defaultzone = fr These are zapata.conf: - [channels] context=inicio switchtype=euroisdn signalling=pri_cpe (pri_net in server#2) rxwink=300; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 signalling=pri_cpe (pri_net in server#2) channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 Server#1 (caller) extensions.conf: --- [general] static=yes writeprotect=no [inicio] exten = s,1,PlayBack(laxana) exten = s,2,Goto(s,1) exten = t,1,hangup exten = i,1,hangup exten = o,1,hangup exten = h,1,hangup Server#2 (callee) extensions.conf - [general] static=yes writeprotect=no [inicio] exten = s,1,Answer exten = s,2,Agi,600agi exten = s,3,hangup exten = t,1,hangup exten = i,1,hangup exten = o,1,hangup exten = h,1,hangup Do you see something weird??? Thnaks people! -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i make best use of Macro?
On Wed, 11 Jun 2003, Tilghman Lesher wrote: On Wednesday 11 June 2003 10:43 am, Christopher Arnold wrote: a) Is there state building up if my macro calls itself recusivly? A macro is NOT a function. It simply is a shortcut to doing a longer series of commands. A macro cannot itself be recursive. Hmm shame on me... So that is why it is called a Macro! :-) Or should i use a Goto(newcontext,s,1)? I think you're confused as to what a macro can and cannot do. Until you understand that distinction, you're going to have trouble understanding how a macro works within the system. Shure i am. So i really should se a macro as something close to a #define in C? In any case, I'd recommend that you forget about using macros and instead use AGI. I had the impression that it was possible without AGI, but ok ill follow the oracles advice. Does anyone have any pointers to documentation on the AGI interface? Another state question: It would be possible to implement my functionality with a circle of contexts. (I actually have a running proof of concept implementation) But how is it in this case, would asterisk build up a huge state if someone rotates around to much in the loop of contexts? /Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy message with call waiting?
Is it possible to have both a busy and an away message when the call waiting feature is enabled? extensions.conf ... exten=403,1,Dial,Zap/3|10 exten=403,2,Voicemail2,u403 exten=403,103,Voicemail2,b403 ... Because I have enabled call waiting, I can't see how it will be possible to get the busy message to play (because there will always be a dial tone). Am I right, or do I have incorrect configurations? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i make best use of Macro?
On Wednesday 11 June 2003 01:01 pm, Christopher Arnold wrote: On Wed, 11 Jun 2003, Tilghman Lesher wrote: On Wednesday 11 June 2003 10:43 am, Christopher Arnold wrote: a) Is there state building up if my macro calls itself recusivly? A macro is NOT a function. It simply is a shortcut to doing a longer series of commands. A macro cannot itself be recursive. Hmm shame on me... So that is why it is called a Macro! :-) Or should i use a Goto(newcontext,s,1)? I think you're confused as to what a macro can and cannot do. Until you understand that distinction, you're going to have trouble understanding how a macro works within the system. Shure i am. So i really should se a macro as something close to a #define in C? Quite analogous. In any case, I'd recommend that you forget about using macros and instead use AGI. I had the impression that it was possible without AGI, but ok ill follow the oracles advice. Does anyone have any pointers to documentation on the AGI interface? http://asterisk.drunkcoder.com/agi.cgi It certainly is possible without AGI, but then again, I tend to code stuff in C even when it probably would be faster to write in Perl and AGI. Another state question: It would be possible to implement my functionality with a circle of contexts. (I actually have a running proof of concept implementation) But how is it in this case, would asterisk build up a huge state if someone rotates around to much in the loop of contexts? As you aren't calling functions, you aren't creating a deeper stack, and therefore, aren't accumulating additional state information. Every branch is just a jump, with no return address stored. Subroutines might be in the future of the Asterisk extension logic, but they aren't there now. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] lost variables
Why do you think so? Local variables get lost only when the call gets hanged up. Martin On Wed, 11 Jun 2003, Paulo Mannheimer wrote: Hi, Seems that my local variable content get lost when I call an AGI program. Is this the correct functionality? Thanks, Paulo H. Mannheimer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i make best use of Macro?
Okay, while reading over this thread it occured to me one more feature that should be real simple to add to app_meetme.c that would solve quite a bit of what is trying to be done here. The feature that needs to be added is a function to pass in a variable and let meetme populate it with the current number of users. Possibly have it be a second argument to the current MeetMeCount that would populate the variable and skip the ast_say_number function. This would allow it to be backwards compatible, while moving forward. How it would benefit the current question is that you could basically script up a fall through set of priorities that made it work exten= x,1,MeetMeCount(1234,num) exten= x,2,GotoIf count is large exten= x,3,MeetMe(1234) exten= x,4,Hangup or otherwise exten= x,5,MeetMeCount(1235,num) exten= x,6,GotoIf Of course as you can see this is the way a Macro would come in handy then because there is a lot of repetition with only the MeetMe number changing. The diff if done to MeetMeCount shouldn't be more than 10 or so lines. If there is interest in it, I'll work on it this afternoon. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i make best use of Macro?
On Wednesday 11 June 2003 02:09 pm, Steven Critchfield wrote:
Okay, while reading over this thread it occured to me one more
feature that should be real simple to add to app_meetme.c that
would solve quite a bit of what is trying to be done here. The
feature that needs to be added is a function to pass in a variable
and let meetme populate it with the current number of users.
Possibly have it be a second argument to the current MeetMeCount
that would populate the variable and skip the ast_say_number
function. This would allow it to be backwards compatible, while
moving forward.
How it would benefit the current question is that you could
basically script up a fall through set of priorities that made it
work
exten= x,1,MeetMeCount(1234,num)
Applications can only take a single argument. I would propose
using something like (e1234) - e for export variable; then export
the value to a common variable like ${MEETMECOUNT}.
exten= x,2,GotoIf count is large
-Tilghman
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Re: [Asterisk-Users] Opportunistic VoIP
This is slightly off-topic I suppose, but: I'd say it's on-topic, since it's something (if implemented) could radically change the way Asterisk moves calls between servers. If understood correctly, I believe it could be the single biggest change that the VOIP industry (movement?) could use to destroy the existing traditional infrastructure of the phone system. Asterisk seems to be a pretty good hammer right now for starting that job; I think TRIP would give some extra muscle to the task. At 20:37 10-6-2003 -0700, you wrote: You should investigate TRIP (RFC 3129): http://www.zvon.org/tmRFC/RFC3219/Output/ Find BSD-licensed proof-of-concept code at http://www.vovida.org/downloads/trip/trip-1.0.0.tar.gz If someone could incorporate this into Asterisk and extend the functionality, that would be pretty nice. The basic ENUM support in Asterisk already can handle specific number paths, but I think TRIP or something like TRIP would be best for handling situations where larger groups of numbers need to be advertised into a routing table behind a particular Asterisk server. Think BGP for phone numbers. I'm sorry, but I see no real benefits to TRIP over ENUM. Large amounts of data in DNS databases have not been a real problem yet, provided the tree is delegated properly (as ENUM does), and works quite effectively due to caching. TRIP only makes it harder for widespread use to deal with such things as number portability (can't ever do that with IP, remember). As far as I can tell from the TRIP docs this looks a lot like some big telco tries to make it more difficult for customers to move to another telco and still use their old number... Florian I see large benefits in using TRIP versus ENUM. I'll list some below, with #1 and #2 being the most important, and the others in no particular order. 1) The ENUM architecture is controlled by national or international governing bodies. Ultimately, they can restrict or charge for data in the ENUM database, and unless you split your root servers, you are stuck with whatever policies, speed of response, and political issues that introduces. This is a _huge_ problem - note that ENUM is not deployed in the US due to political issues, and not technical ones. How do you feel about paying Verisign for your phone number? 2) The ENUM system is centralized. TRIP can be established between two telephone systems, independently of any third party's cooperation or assistance. Routes can be exchanged in any way that is acceptable to those two systems. 3) ENUM is DNS based, and is subject to the delays, trials and tribulations of that protocol. TRIP is based on peer-to-peer TCP sessions which flood updates to each other, and architecturally can handle changes to the route table more quickly (though still not ideal.) 4) ENUM is really designed to answer specific questions about individual numbers, and it has exactly one set of answers for those particular numbers. TRIP is designed for aggregating number prefixes in route-like formats. This allows overlap and competition between servers that may be offering the same path. TRIP allows the use of alternate values (communities and preferences, as well as extendable features in the attributes fields) that allow decision-making on destination choices. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] filling suppressed silence with chan_oh323
After some more analysis of my dropped fragment problem, things look like this: Cisco 7940 phone -- RTP -- chan_oh323 -- Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager 3.2, but this part should not be relevant. Connections work fine, with one exception: Whenever there's a break in *'s voice stream (eg. between the mailbox and password prompts), the 7940 detects horrible jitter and drops a few packets (eg. the whole password prompt). Using ethereal, I found that the RTP packets sent by asterisk seem to have bogus timestamps: After the gap, timestamps continue just as if there hasn't been a gap, so timestamp / sequence number always is constant. This should be fine for continuous RTP streams, so I tried disabling silence suppression in oh323.conf. However, * still only sends out packets while it is playing, and not between playback phases. So AFAICT, there are two possible solutions: 1) make chan_oh323 stream continuously, no matter if the current application does not play audio. IOW: transmit silence instead of no packets. Is this possible? 2) use better timestamps in streamed packets, ie increase timestamps even after a period of silence, and not only for each sent packet. Not sure if that makes the phone happy, though... Any chance to do one of those? Thanks in advance, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i make best use of Macro?
On Wed, 2003-06-11 at 14:43, Tilghman Lesher wrote:
On Wednesday 11 June 2003 02:09 pm, Steven Critchfield wrote:
Okay, while reading over this thread it occured to me one more
feature that should be real simple to add to app_meetme.c that
would solve quite a bit of what is trying to be done here. The
feature that needs to be added is a function to pass in a variable
and let meetme populate it with the current number of users.
Possibly have it be a second argument to the current MeetMeCount
that would populate the variable and skip the ast_say_number
function. This would allow it to be backwards compatible, while
moving forward.
How it would benefit the current question is that you could
basically script up a fall through set of priorities that made it
work
exten= x,1,MeetMeCount(1234,num)
Applications can only take a single argument. I would propose
using something like (e1234) - e for export variable; then export
the value to a common variable like ${MEETMECOUNT}.
True and false. You get passed data, and what you do with data is up to
you. I could split on some non important character and do what I want
with the string. So while you are correct, I can do anything I want with
my string.
--
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[Asterisk-Users] Voicemail notification
Besides email notification, is there another way to get asterisk notify the user that they have a message? Example: Some analog phones have a blinking light that lets the user know that they have a voicemail message. Is asterisk capable of doing this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems configuring Asterisk with SIP
Hi everybody Could someone give a tip on how can I configure asterisk to use 2 ATA's 186 to communicate each other using SIP with asterisk. I know this most be a very simple task, however this is the very first aproach I have to asterisk. I set the following config but I don't get dial-tone when I off-hook the phone from any of the two ATAs. Can some one tell what I'm missing in the configuration?? sip.conf file [general] port = 5060 ; Port to bind to bindaddr = 192.168.0.254 ; Address to bind to context = default ; Default for incoming calls tos=lowdelay ;tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ; ;register = [EMAIL PROTECTED] ; Register with a SIP provider ;register = [EMAIL PROTECTED] ;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as 1234 here. ;allow=g729 ; ;[cisco] type=friend username=9873 secret=pwd ;nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away defaultip=192.168.0.5 mailbox=9873 ;[cisco2] type=friend username=9874 secret=pwd nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away defaultip=192.168.0.10 mailbox=9874 extensions.conf I added this at the end of the extension.conf file: exten = 9873,1,Dial(SIP/cisco,30,tr) exten = 9873,2,Playback(new/nbdy-avail-to-take-call) exten = 9873,3,Voicemail(u9873) exten = 9873,4,Hangup exten = 9873,102,Voicemail(b9873) exten = 9873,103,Hangup exten = 9874,1,Dial(SIP/cisco2,30,tr) exten = 9874,2,Playback(new/nbdy-avail-to-take-call) exten = 9874,3,Voicemail(u9874) exten = 9874,4,Hangup exten = 9874,102,Voicemail(b9874) exten = 9874,103,Hangup And that's all I did. However I'm not sure If I have to configure something else?? I also have a SIP proxy server(Not asterisk) and I pretend to send out calls through this proxy server, but thisonce the 2 ATAs can call to each other behid asterisk. Can someone give a hint on this??? Any tip would be appreciated. I'm actually using Redhat 9. The ATAs are using the 2.16 firmware. The ATA's are pointing to the asterisk bind address 192.168.0.254 in the sip.conf Thanks in advance!! Kind Regards!! This is what I have: ATA 1, UID0=9873 192.168.0.5- | |--Asterisk BOX-SIP-Proxy Server ATA 2, UIDO=9874 | 192.168.0.254 192.168.0.2 192.168.0.10 ---
Re: [Asterisk-Users] Opportunistic VoIP
At 12:58 11-6-2003 -0700, you wrote: I see large benefits in using TRIP versus ENUM. I'll list some below, with #1 and #2 being the most important, and the others in no particular order. 1) The ENUM architecture is controlled by national or international governing bodies. Ultimately, they can restrict or charge for data in the ENUM database, and unless you split your root servers, you are stuck with whatever policies, speed of response, and political issues that introduces. This is a _huge_ problem - note that ENUM is not deployed in the US due to political issues, and not technical ones. How do you feel about paying Verisign for your phone number? Sure, this is true. However, if no widely acceptable ruleset is defined, alternative roots may rise (who says enum MUST be applied below e164.arpa ?). 2) The ENUM system is centralized. TRIP can be established between two telephone systems, independently of any third party's cooperation or assistance. Routes can be exchanged in any way that is acceptable to those two systems. See 1) There is no reason to not run ENUM on other zones for 'private' use. 3) ENUM is DNS based, and is subject to the delays, trials and tribulations of that protocol. TRIP is based on peer-to-peer TCP sessions which flood updates to each other, and architecturally can handle changes to the route table more quickly (though still not ideal.) I agree that this is a great way to deal with blocks of numbers, just like it is a great way to deal with blocks op IP-adresses. However, as BGP sucks in routing huge amounts of singular numbers, I expect TRIP to suck at routing huge amounts of individual phonenumbers. This is an issue I need to deal with for an ongoing project myself, and I'm not seeing how its adressed in TRIP. 4) ENUM is really designed to answer specific questions about individual numbers, and it has exactly one set of answers for those particular numbers. TRIP is designed for aggregating number prefixes in route-like formats. This allows overlap and competition between servers that may be offering the same path. TRIP allows the use of alternate values (communities and preferences, as well as extendable features in the attributes fields) that allow decision-making on destination choices. Hmm, now this may have use, however, the same effect is reached by implementing this on an IP-level (in BGP as opposed to in TRIP), or isn't it ? Don't get me wrong - I have no need to burn down TRIP or elevate ENUM. I am just trying to figure out each respective value for future telephony. Thanks for your comments! Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NewbieQ: SOHO setup with x100p
On Wed, Jun 11, 2003 at 10:12:22AM +0200, Tielman Koekemoer wrote: On Tue, 10 Jun 2003, Tielman Koekemoer wrote: welcome to * Good to be here. c) get a T400P + channel bank (expensive, but it does give you 24 ports) I'm also considering a PRI from our local Telco (thanks to Mr Davies)connected to an E100P but am waiting for a quote from said Telco to compare costs with your idea (T[1,4]00P + channel bank). If any, what are the advantages of either? Off the cuff please, I don't expect lengthy answers. Would I need a channel bank with the E100P? I'm asking as I need to keep costs low. Getting the lines delivered over a PRI would eliminate the need for the channel bank for FXO ports. If you want to use non-VoIP phones, you still need some way to deal with FXS ports. But I think you intended to use soft phones internally so that wouldn't be a problem for you. You would just plug the E1 into the E100P and be done with it. -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Only noise in zap channel
On Wed, Jun 11, 2003 at 11:36:26AM -0300, Eduardo Goncalves wrote: On Tue, 10 Jun 2003 14:36:25 -0400 Scott Lambert [EMAIL PROTECTED] wrote: Is the noise loud and sounds like you have picked up the phone in the middle of a modem call? If so, I had a similar problem with my TDM20 while it was sharing an IRQ with the unused AC97 chip. I shuffled the cards around to different PCI slots and it now works. In my problem, in addition to the noise, asterisk was not responding to DTMF tones pressed on the analog handset. I disabled all of the cards on my board (serial, paralel, etc) but the problem remains. I can't hear, but the person on the other side can hear me. That is different from my problem. In my case, the noise was on both ends of the conversation. -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hardware - Channelbank vs SIP etc
On Wed, Jun 11, 2003 at 12:42:57AM -0500, denon wrote: We're doing a new * installation at a remote office soon, and I was just curious what people's opinions were on hardware these days .. I've had decent luck with T100Ps and Adtran, but I know times change .. I'm looking to do roughly 15 handsets and 15 pstn, with some room to grow. I had planned on two T100Ps and two adtran 750s, one for handsets, one for pstn. You might check the pricing on getting your 15 lines delivered on T1/PRI. It may be the same price or even cheaper. And you save one channel bank and the associated complexities. It also only take 2 pair; which can be important in some neighborhoods. flashbacks of taking out an entire CO the first night we brought up our new PoP with 300 POTS dail-up lines. The telco thanked us when we put in Cisco AS5200s instead of our Lucent PM2es. I'm thinking of going SIP on the other side, though. I've been looking at the Grandstream budgetone phones, as well as their handytone. Anyone have anything good or bad to say on these? Cisco is out of that office's budget, I'm afraid. We're replacing a cheapo key system there, so it's all about the benjamins.. :\ I was also looking at: http://clipcomm.co.kr/eng/e_product/e_product_voip_analoggateway_4.html (rumored to be D-Link's OEM?) and http://www.yoda.com.tw/SOLUTIONS/vg422r.htm Any thoughts on these? Has anyone had good luck with other low-cost channels banks? (noo, not Zhone.. :) Any tips are appreciated, you can catch me here or on irc as always .. -d ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI and SET VARIABLE
I am having a problem understanding/visualizing the environment of AGI and how
variables defined there can be used in my dial plan. I am so close I can
taste it. I just want to return a number to dial from a list of numbers in a
file.
from extensions.conf
[talk2doc]
; Please Hold While I Transfer Your Call
exten = s,1,AGI(pnumber.agi)
exten = s,2,Dial(Zap/2/$[PHONE_NUM]|15)
in my agi perl script - pnumber.agi
.
if ( $cntr = $#file ) {
print SET VARIABLE PHONE_NUM $file[$cntr - 1];
$cntr = 1;
}
else {
print SET VARIABLE PHONE_NUM $file[$cntr - 1];
$cntr++;
}
If I open up a CLI and dial up asterisk and press the appropriate extension I
can see it run the agi script but no returned var.
-- Goto (talk2doc,s,1)
-- Executing AGI(Zap/1-1, pnumber.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/pnumber.agi
-- AGI Script pnumber.agi completed, returning 0
-- Executing Dial(Zap/1-1, Zap/2/$PHONE_NUM|15) in new stack
-- Called 2/$PHONE_NUM
--
Mark Street, D.C.
Red Hat Certified Engineer
Cert# 807302251406074
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[Asterisk-Users] Telephone Tree
Hi everyone, I'd like to use Asterisk to build a phonetree (www.phonetree.com) type of application, like this: 1. Read a text-based name/phonenumber file. 2. Call every number and play a recorded message. 3. If a beep is detected, replay the message from scratch (to leave messages on an answering machine). 4. Write results to a log file. Does anything like this exist already? Can this be done with Asterisk's script syntax? Thanks, Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI and SET VARIABLE
Why bother returning the value when you can just dial directly from AGI.
On Wed, 2003-06-11 at 18:35, Mark Street wrote:
I am having a problem understanding/visualizing the environment of AGI and how
variables defined there can be used in my dial plan. I am so close I can
taste it. I just want to return a number to dial from a list of numbers in a
file.
from extensions.conf
[talk2doc]
; Please Hold While I Transfer Your Call
exten = s,1,AGI(pnumber.agi)
exten = s,2,Dial(Zap/2/$[PHONE_NUM]|15)
in my agi perl script - pnumber.agi
.
if ( $cntr = $#file ) {
print SET VARIABLE PHONE_NUM $file[$cntr - 1];
$cntr = 1;
}
else {
print SET VARIABLE PHONE_NUM $file[$cntr - 1];
$cntr++;
}
If I open up a CLI and dial up asterisk and press the appropriate extension I
can see it run the agi script but no returned var.
-- Goto (talk2doc,s,1)
-- Executing AGI(Zap/1-1, pnumber.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/pnumber.agi
-- AGI Script pnumber.agi completed, returning 0
-- Executing Dial(Zap/1-1, Zap/2/$PHONE_NUM|15) in new stack
-- Called 2/$PHONE_NUM
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] Telephone Tree
On Wednesday 11 June 2003 08:08 pm, Dylan VanHerpen wrote: Hi everyone, I'd like to use Asterisk to build a phonetree (www.phonetree.com) type of application, like this: 1. Read a text-based name/phonenumber file. 2. Call every number and play a recorded message. 3. If a beep is detected, replay the message from scratch (to leave messages on an answering machine). 4. Write results to a log file. Does anything like this exist already? Can this be done with Asterisk's script syntax? Thanks, Dylan. I hope it's not one of those horrible automated marketing machines you are building... -- __ This sig is pending approval ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail notification
it should be added to zapata.conf, and you can specify multiple mailboxes separated by , On Wed, 2003-06-11 at 20:10, Andy Powell wrote: I'd like to use either the message waiting light or stutter tone but on searching the archives I found conflicting answers. Everyone seems to agree that you should add mainbox=mailbox number but some people are saying that it should be added to zapata.conf and others are saying zaptel.conf Can someone who has it working clarify this? If it is zaptel.conf can somone supply a sample.. my zaptel.conf file only consists of fxsks=1 fxoks=2 fxoks=3 loadzone=uk defaultzone=uk and that's it... Thanks in advance Andy *** REPLY SEPARATOR *** On 11/06/2003 at 16:53 Steven Critchfield wrote: On Wed, 2003-06-11 at 15:16, Derek Beaumont wrote: Besides email notification, is there another way to get asterisk notify the user that they have a message? Example: Some analog phones have a blinking light that lets the user know that they have a voicemail message. Is asterisk capable of doing this? Yes, and I know it works on Sip and Zap channels. Check archive for MWI, for Message waiting indicator. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI and SET VARIABLE
On Wednesday 11 June 2003 17:10, Steven Critchfield wrote:
Why bother returning the value when you can just dial directly from AGI.
Because my feeble mind is being streched a bit by AGI. Throw me a bone
man. I downloaded and installed the asterisk-perl modules and changed my
script to use those. The docs are not clear on how to dial using the AGI
class to dial out.
I corrected some errors in my syntax in extensions.conf... Nice output from
the agi script from command line but when * is called CLI shows no data in my
var... so close but yet so far
Goto (talk2doc,s,1)
-- Executing AGI(Zap/1-1, pnumber.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/pnumber.agi
-- AGI Script pnumber.agi completed, returning 0
-- Executing Dial(Zap/1-1, Zap/2/|15) in new stack
-- Called 2/
-- Zap/2-1 answered Zap/1-1
-- Attempting native bridge of Zap/1-1 and Zap/2-1
-- Hungup 'Zap/2-1'
--
in my pnumber.agi script I set;
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
if ( $cntr = $#file ) {
#print SET VARIABLE PHONE_NUM $file[$cntr - 1];
$AGI-set_variable('PHONE_NUM', $file[$cntr - 1]);
$cntr = 1;
}
else {
$AGI-set_variable('PHONE_NUM', $file[$cntr - 1]);
#print SET VARIABLE PHONE_NUM $file[$cntr - 1];
$cntr++;
}
from extensions.conf
[talk2doc]
; Please Hold While I Transfer Your Call
exten = s,1,AGI(pnumber.agi)
exten = s,2,Dial(Zap/2/${PHONE_NUM}|15)
--
Mark Street, D.C.
Red Hat Certified Engineer
Cert# 807302251406074
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Re: [Asterisk-Users] Telephone Tree
Steve wrote: On Wednesday 11 June 2003 08:08 pm, Dylan VanHerpen wrote: Hi everyone, I'd like to use Asterisk to build a phonetree (www.phonetree.com) type of application, like this: 1. Read a text-based name/phonenumber file. 2. Call every number and play a recorded message. 3. If a beep is detected, replay the message from scratch (to leave messages on an answering machine). 4. Write results to a log file. Does anything like this exist already? Can this be done with Asterisk's script syntax? Thanks, Dylan. I hope it's not one of those horrible automated marketing machines you are building... Well, I guess you'd have to include a disclaimer not to use it for marketing or political purposes ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telephone Tree
On Wed, Jun 11, 2003 at 07:44:47PM -0600, Dylan VanHerpen wrote: Well, I guess you'd have to include a disclaimer not to use it for marketing or political purposes ;) Perhaps an 'abuse' clause is needed. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail notification
On Wednesday 11 June 2003 19:10, Andy Powell wrote:
I'd like to use either the message waiting light or stutter tone but
on searching the archives I found conflicting answers.
Everyone seems to agree that you should add
mainbox=mailbox number
but some people are saying that it should be added to zapata.conf and
others are saying zaptel.conf
Can someone who has it working clarify this?
zaptel.conf is used for the kernel module. zapata.conf is used for the
Asterisk program. As MWI is an Asterisk feature, it must therefore be
placed in zapata.conf.
If you're still not convinced, you can do a grep in the various cvs
repositories to confirm which is which:
[EMAIL PROTECTED]:/cvs/asterisk# grep -r 'mailbox' /cvs/zaptel
[EMAIL PROTECTED]:/cvs/asterisk# grep -r 'mailbox' channels/chan_zap.c
} else if (!strcasecmp(v-name, mailbox)) {
} else if (!strcasecmp(v-name, mailbox)) {
-Tilghman
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RE: [Asterisk-Users] Using Linux traffic shaping to prioritise SIP/IAX traffic?
Hi Alberto, Being a QOS newbie, your example was invaluable! I'm testing your example, and so far so good. Once I have something I'm happy with, I'll post it on my Asterisk website: http://www.wwworks-inc.com/asterisk Nice work Alberto, and thanks. -wade On Tue, Jun 10, 2003 at 09:58:14PM +0200, Emanuele Pucciarelli wrote: Il mar, 2003-06-10 alle 20:07, Stephen Davies ha scritto: When the tos option is set correctly (to nodelay), the default queueing in recent kernels already does that, because the pfifo_fast queue is used (if I recall correctly). But there is never any queue on my Linux box. It all storms out of the ethernet interface and gets queued up in my cable modem which doesn't know anything about tos settings. That is not entirely correct. There is an output queue, and pfifo_fast is the default (see the LARTC Howto, 9.2.1.1). But you are right when you say you need something to slow down the data;the simplest choice should be addingthe Token Bucket Filter (9.2.2.2). But if the wondershaper already does it all, then it's probably better to go with it... :) You should read further into lartc.org, the linux traffic shaping capabilities are really wide and you can find lots of ways of doing what you want. In your case, I guess the logical choice would be to use HTB, with two classes, let's say if your cablemodem is 512kbps, you can save 112kbps for voice and signalling (yes it's extreme but it's an example), and 400kbps for the data. Also, you can put the former with top priority to minimize latency. Under those you can use sfq to make everything more fair, that helps a lot when saturating. This is a very short script, I'm sure there is something like this on lartc or htb's website. Something like this (completely untested, from memory so don't trust me): # delete the existing qdisc tc qdisc del dev eth1 root 21 /dev/null # init the htb qdisc tc qdisc add dev eth1 root handle 1: htb tc class add dev eth1 parent 1: classid 1:5 htb rate 512kbit tc class add dev eth1 parent 1:5 classid 1:10 htb rate 112kbit prio 0 tc class add dev eth1 parent 1:5 classid 1:20 htb rate 400kbit prio 2 # sfq for all of them tc qdisc add dev eth1 parent 1:5 handle 500: sfq perturb 10 tc qdisc add dev eth1 parent 1:10 handle 100: sfq perturb 10 tc qdisc add dev eth1 parent 1:20 handle 200: sfq perturb 10 And now you need to set the filters up, which can be based on tc or using iptables' MARK target. For instance: # everything marked 10 in iptables go to 1:10 (the 112kbit) tc filter add dev eth1 protocol ip parent 1:0 prio 1 handle 10 fw \ flowid 1:10 # everything marked 20 in iptables go to 1:20 (the 400kbit) tc filter add dev eth1 protocol ip parent 1:0 prio 1 handle 20 fw \ flowid 1:20 And then mark in iptables (i don't remember sip's port very well, and you should also add RTP stuff too): # sip, mark 10 iptables -t mangle -A PREROUTING -i eth2 -p tcp --destination-port 5060 \ -j MARK --set-mark 10 iptables -t mangle -A PREROUTING -i eth2 -p udp --destination-port 5060 \ -j MARK --set-mark 10 # default, mark 20 iptables -t mangle -A PREROUTING -i eth2 -d 0.0.0.0/0 \ -j MARK --set-mark 20 In this case you could have used tc's native filters which are much faster than iptables, but also harder to setup, so if this is your first approach to this stuff I wouldn't recommend them (and don't worry, the speed difference is _not_ noticeable for that bandwidth). I hope this helps, however this is all untested, (ie. just wrote it) so please look into the docs to find out more. Thanks, Alberto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI and SET VARIABLE
Notice that you should refer to PHONE_NUM variable this way:
${PHONE_NUM}
Martin
On Wed, 11 Jun 2003, Mark Street wrote:
I am having a problem understanding/visualizing the environment of AGI and how
variables defined there can be used in my dial plan. I am so close I can
taste it. I just want to return a number to dial from a list of numbers in a
file.
from extensions.conf
[talk2doc]
; Please Hold While I Transfer Your Call
exten = s,1,AGI(pnumber.agi)
exten = s,2,Dial(Zap/2/$[PHONE_NUM]|15)
in my agi perl script - pnumber.agi
.
if ( $cntr = $#file ) {
print SET VARIABLE PHONE_NUM $file[$cntr - 1];
$cntr = 1;
}
else {
print SET VARIABLE PHONE_NUM $file[$cntr - 1];
$cntr++;
}
If I open up a CLI and dial up asterisk and press the appropriate extension I
can see it run the agi script but no returned var.
-- Goto (talk2doc,s,1)
-- Executing AGI(Zap/1-1, pnumber.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/pnumber.agi
-- AGI Script pnumber.agi completed, returning 0
-- Executing Dial(Zap/1-1, Zap/2/$PHONE_NUM|15) in new stack
-- Called 2/$PHONE_NUM
--
Mark Street, D.C.
Red Hat Certified Engineer
Cert# 807302251406074
--
Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0
GPG key http://www.streetchiro.com/pubkey.asc
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Re: [Asterisk-Users] How do i make best use of Macro?
Since there was some interest in this, here is the diff against current
cvs. Someone that is better at C should look into my use of strsep
because there is a couple of warnings. Also there is a warning on my use
of pbx_builtin_setvar_helper, but I can't see whats wrong here.
BTW, SayNumber doesn't seem to say '0'.
Usage is like this.
exten = 1234,1,MeetMeCount(1234|var)
exten = 1234,2,SayNumber(${var})
exten = 1234,3,MeetMe(1234)
-
diff -U3 -r asterisk-orig/apps/app_meetme.c asterisk/apps/app_meetme.c
--- asterisk-orig/apps/app_meetme.c 2003-06-11 23:14:38.0 -0500
+++ asterisk/apps/app_meetme.c 2003-06-11 22:58:32.0 -0500
@@ -54,9 +54,10 @@
'q' -- quiet mode (don't play enter/leave sounds)\n;
static char *descrip2 =
- MeetMeCount(confno): Plays back the number of users in the specified MeetMe\n
-conference. Returns 0 on success or -1 on a hangup. A ZAPTEL INTERFACE\n
-MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY.\n;
+ MeetMeCount(confno[|var]): Plays back the number of users in the specifiedi\n
+MeetMe conference. If var is specified, playback will be skipped and the value\n
+will be returned in the variable. Returns 0 on success or -1 on a hangup.\n
+A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY.\n;
STANDARD_LOCAL_USER;
@@ -465,19 +466,29 @@
int res = 0;
struct conf *conf;
int cnt;
+ char* confnum;
+ char val[5] = 0; /* I don't think we will ever get 99,999 callers into a
single meetme */
+
if (!data || !strlen(data)) {
ast_log(LOG_WARNING, MeetMeCount requires an argument (conference
number)\n);
return -1;
}
LOCAL_USER_ADD(u);
- conf = find_conf(data, 0);
+ confnum = strsep((char*) data,|);
+ conf = find_conf(confnum, 0);
if (conf)
cnt = conf-users;
else
cnt = 0;
- if (chan-_state != AST_STATE_UP)
- ast_answer(chan);
- res = ast_say_number(chan, cnt, , chan-language);
+ if(strlen(data)){
+ /* have var so load it and exit */
+ sprintf(val,%i,cnt);
+ pbx_builtin_setvar_helper(chan,(char*) data,val);
+ }else{
+ if (chan-_state != AST_STATE_UP)
+ ast_answer(chan);
+ res = ast_say_number(chan, cnt, , chan-language);
+ }
LOCAL_USER_REMOVE(u);
return res;
}
--
Steven Critchfield [EMAIL PROTECTED]
___
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Re: [Asterisk-Users] Testing two E400P with E1 cross-cable
I'm using a self-made cable with just 4-wires to hook up 2 E100P. Has been working for few months without trouble. Just connect: Pin 1 to Pin 4 Pin 2 to Pin 5 Pin 4 to Pin 1 Pin 5 to Pin 2 It's the same as a T1 crossover cable. IIRC, the E1/T1 sends signal out on pair 4/5 and receives on 1/2. So, crossing the pairs should be correct. Some Cisco docs also described the same thing (for connecting their voice routers to your PBX). Regarding the grounding or so called shielding, I was told it has to be properly done according to telco specs. In fact, some E1 vendors will explicitly state that you should get someone qualified to make the cable if it involves the grounding. So far, I've been running fine without the grounding. I think it's ok if the distance is short, but if the distance is far I think you're better off with proper grounding. It would be helpful if someone can help clarify this. My gut feel: it's probably very important if you've ever seen the thickness of ground cables used in the COs. Cheers. Carlos Cars wrote: Jared Smith escribi: I have a funny feeling your crossover cable might be wrong... I'm not sure about an E1 crossover, but I know that a T1 crossover is different than a standard ethernet crossover. (See http://www.jaredsmith.net/misc/cables/) If you do find the pinout for an E1 crossover, let me know and I'll add it to my site. Jared Smith Right now I'm testing this pinout: pin1 (Rx -) -- pin4 pin2 (Rx +) -- pin5 pin3 (Rx Shield) -- pin3 pin4 (Tx -) -- pin1 pin5 (Tx +) -- pin2 pin6 (Tx Shield) -- pin6 pin7 (not used) -- pin7 pin8 (not used) -- pin8 I don't know if this one is the good one, but zttool says it's ok. I'm not sure, but E1 and T1 cables should be the same... Best Regards, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Opportunistic VoIP
At 12:58 11-6-2003 -0700, you wrote: I see large benefits in using TRIP versus ENUM. I'll list some below, with #1 and #2 being the most important, and the others in no particular order. 1) The ENUM architecture is controlled by national or international governing bodies. Ultimately, they can restrict or charge for data in the ENUM database, and unless you split your root servers, you are stuck with whatever policies, speed of response, and political issues that introduces. This is a _huge_ problem - note that ENUM is not deployed in the US due to political issues, and not technical ones. How do you feel about paying Verisign for your phone number? Sure, this is true. However, if no widely acceptable ruleset is defined, alternative roots may rise (who says enum MUST be applied below e164.arpa ?). A widely acceptable ruleset will eventually arise, but why build parallel structures? 2) The ENUM system is centralized. TRIP can be established between two telephone systems, independently of any third party's cooperation or assistance. Routes can be exchanged in any way that is acceptable to those two systems. See 1) There is no reason to not run ENUM on other zones for 'private' use. Historically, attempts at alternate roots have failed, as they should. The DNS should remain cohesive. While using alternate roots, or private zones, may work internally within organizations, this rapidly falls apart when crossing autonomous boundaries. How do you filter particular records from certain resolvers? How do you get granular control over even your own zones without re-writing BIND to support these new methods? ENUM is great for individual numbers, but again, it is not apparent to me how it is going to be useful for range-based announcements or how it will be used with any weighting mechanism that is determined by the end owner of the route. 3) ENUM is DNS based, and is subject to the delays, trials and tribulations of that protocol. TRIP is based on peer-to-peer TCP sessions which flood updates to each other, and architecturally can handle changes to the route table more quickly (though still not ideal.) I agree that this is a great way to deal with blocks of numbers, just like it is a great way to deal with blocks op IP-adresses. However, as BGP sucks in routing huge amounts of singular numbers, I expect TRIP to suck at routing huge amounts of individual phonenumbers. This is an issue I need to deal with for an ongoing project myself, and I'm not seeing how its adressed in TRIP. The current problems with BGP are relevant to global route tables. I suspect that aggregation on a much larger scale is possible with phone numbers. Even if it is not, the route selection methods of TRIP are easily extracted to alternate processing methods, and commodity hardware is _cheap_. BGP tables are a crisis because of expensive vendor-specific hardware requirements (and even now, they're really not a crisis - the Internet works.) By the time that tens of thousands of companies are putting their phone systems into TRIP-capable networking meshes, a gigabyte of RAM in a machine will be standard. I am not worried at all about scaleability from that angle. 4) ENUM is really designed to answer specific questions about individual numbers, and it has exactly one set of answers for those particular numbers. TRIP is designed for aggregating number prefixes in route-like formats. This allows overlap and competition between servers that may be offering the same path. TRIP allows the use of alternate values (communities and preferences, as well as extendable features in the attributes fields) that allow decision-making on destination choices. Hmm, now this may have use, however, the same effect is reached by implementing this on an IP-level (in BGP as opposed to in TRIP), or isn't it ? Don't get me wrong - I have no need to burn down TRIP or elevate ENUM. I am just trying to figure out each respective value for future telephony. As am I. I'm simply trying to be pragmatic. I have a number of customers, all of whom require long distance service from a provider. They all run Asterisk. I would like to be able to create a TRIP peer between my customer and five long distance providers (after paying the account signup fee, of course) and then have the routing system start to choose which provider it's going to use. Maybe some providers don't have service to some countries - so, I simply would not see those country codes in the route tables from those providers. I might make my choices based completely on price as a metric, with the best provider winning and the others as backup. Then, as I get more sophisticated, I can start to weigh certain area codes as having better quality on some providers, even though they're more expensive, and I can start to shift my traffic for certain circumstances over to that provider. And then, let's say that one of my clients
[Asterisk-Users] Thank you very much
To James, Robert, Woody, and last but not least, Leo. Thank you very much for your suggestions on Zaurus mic/headphone configurations and the link for the softphone apps. Your help is much appreciated. Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
