Re: [Asterisk-Users] Using a TE405P to connect to an existing PBX
Somewhat. You got the remote site right. I have several Voice T1's at my main location, and it runs into a Cisco router which converts it to SIp and sends it to Asterisk. I would like to be able to push certain incoming phone numbers across IAX to another Asterisk server at a remote site. There, it would send it down a Virtual T1 into the existing PBX system. If this works, I could get another Asterisk server to do the same thing, to simulate the 2nd T1, in case they go over the 23 phone lines on the 1st one. Can this be done? At 02:48 AM 8/14/2004, you wrote: On Fri, 13 Aug 2004, drodden wrote: Sorry guys. Wrong model, I meant a T100P card. I have 1, and can order another if we get this one working. I could use 1 server, or 2. Sorry, I am a little confused on the lingo. I understand that what we have is 3 T1 PRI links, from 3 different carriers. That's how it was described to me. There are 23 phone lines and 1 Data Channel, the B Chan, thus it supports Caller ID and such. We have an old old old ATT voice T1 that has 24 phone lines, no Data channel, thus no caller ID. The PBX we're trying to interface to has 2 T1 PRI's plugged into it. It sounds like the one with 23B+1D is a PRI with ccs, the other one is a digital trunk line then. It is possible to configure the later to have callerid. [snip] We have two of these at the other location. I want to be able to emulate this with 1 or two T100P cards, in 1 or 2 different servers, depending on how much redundancy that location wants. Is this what you want to build? +-local site--++-remote sites---+ /-T1 PRI-\ asterisk --T1 PRI-- remote pbx pbxasterisk --ip- \-T1 PRI-/ asterisk --T1 PRI-- second remote pbx Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound Free World Dialup - extension not ringing?
Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux But I am learning fast. My config is quite simple, Im just following examples and the Wiki: I have two PCs running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to set up Asterisk to accept calls from FWD on another number I have registered, but I cant get my local X-Lite to ring on an inbound call from FWD, and I get the busy tone on the BT100 When I sip debug, I can see that I am registered with FWD, and when I call the number from the BT100 I can see all the incoming information but still nothing on my X-Lite. My extensions.conf: [general] static=yes writeprotect=no [globals] [sip] exten = 1,1,Dial(SIP/phone1,20,tr) exten = 2,1,Dial(SIP/phone2,20,tr) exten = 2,2,VoiceMail,u1234 exten = 2,102,VoiceMail,b1234 ;exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr) exten = 1001,1,Ringing exten = 1001,2,Wait(2) exten = 1001,3,VoicemailMain,s1234 exten = 6601,1,WaitMusicOnHold(60) exten = 232999,1,Dial(SIP/phone1,30,tr) exten = 232999,2,Hangup I am behind a NATed fire wall, but Im not sure that is related. Any ideas or help (working simple confs) would be much appreciated. Best regards -- Chris Blunt SIP: [EMAIL PROTECTED]
[Asterisk-Users] Do you speak Czech?
Continuing the work done on Internationalization of Asterisk, we've begun working on Czech. If you speak Czech, we need you help in continuing the work that has been started. Roll up your sleaves and visit http://bugs.digium.com/bug_view_page.php?bug_id=0002013 Thank you! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004, Peter Svensson wrote: On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: I'll most likely use a BRI. Do you think this will help to avoid echo? Using a BRI will eliminate echos from the pstn connection. Not necessarily! When you call an analog phone via isdn, the other end will introduce echo so that the ip side will be hearing himself speaking with a small delay. I have that problem with my home BRI running zaphfc. Regards, Tobias Jönsson, Lund SE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_oh323 loading error
I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what that mean? Regards Krystian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vlan question
There is a way to ensure traffic prioritisation...but it can work out a little
expensive.
1. Use 3Com 4400 PWR as your switch.
2. Use 3Com NJ200/NJ220 (US) or NJ205/NJ225 (EU) POE Multiport switches
3. Use 3CNJVOIP-CPOD POE -- 7960 POE/Data splitters for power and data connections
to the phone.
The 4400 Delivers Power to the NJ2xx switches, these switches have 4 ports which can
be individually selected to handle traffic prioritisation on a per port basis.
The NJ2xx switches are fully managed supporting SNMP, Power forwarding and 802.1x
security
This is how I've configured my LAN, and Phones get priority for network traffic.
Have fun
P
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: 13 Aug 04, 2:38 PM
Subject: [Asterisk-Users] Vlan question
Hi,
I am setting up an Asterisk system with Cisco 7960 phones. I have a PoE injector to
insert between the patch panel and HP 2626 switch. I plan to plug the users pc into
the phone and the phone into the wall. I would like the phones to have a seperate
subnet from the phones for performance reasons.
May be a silly question, but with the pc and phone sharing the same switch port, how
will it know to seperate the traffic and subnets?
Thanks
tm
ID:[{20040813172548.30403.1811044938-12.6.18.86}]
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Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?
Thanks!!! That works. A bit cumbersome but at least it works. Why does putting someone on hold and then taking them off again make the ATA (or *) recognize the # key as transfer? Thankyou for your help On Sun, 15 Aug 2004 11:49:50 +0800, MPlus [EMAIL PROTECTED] wrote: For blind transfers, press flash twice, then press #. For consultative transfers, press flash once, talk to the other party and tells him to hangup, press flash again, then press #. - Original Message - From: Dennis Cartier [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 15, 2004 2:10 AM Subject: Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm? Unfortunately the Cisco ATA-186 does not support iLBC which means extra costs for purchasing 729 licenses. The ATA-286 works fine other than this 1 issue. Do the Grandstream developers follow this list?? This problem has been persistent for a LONG time and each new firmware version still has it unfixed!! On Sat, 14 Aug 2004 19:36:43 +0300, Lubomir Christov [EMAIL PROTECTED] wrote: Yes, we are experiencing the same problem and because of that we switched the called HT ata to Cisco ATA 186 ... Lubo Andy Lee wrote: [ Message pasted from Sun, 7 Mar 2004 08:27:54 -0600 ] The problem I'm experiencing with many GS adapters, regardless of firmware version is this. Call from one phone to another phone using both the 'T' and 't' flags in the Dial() command. After they are connected, you should be able to press '#' on either phone to hear transfer. What I am experiencing is the calling GS adapter will hear transfer when they press '#', but when the receiving GS adapter presses '#', nothing happens at all. Are you able to repeat this? If not, can you please tell me the firmware revisions and Asterisk version that you are using? Thank you very much. Yes, I am experiencing the same problem. Works fine with the BudgetTone phones but not with the HandyTone-286. Have you resolved this yet? We've been trying to figure this out on and off for the past couple of weeks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Appradius Project: RADIUS authentication and accounting support for Asterisk PBX http://appradius.minitelecom.org/ - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 12:15:20 +0200 (CES), Tobias Jönsson [EMAIL PROTECTED] wrote: On Sun, 15 Aug 2004, Peter Svensson wrote: On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: I'll most likely use a BRI. Do you think this will help to avoid echo? Using a BRI will eliminate echos from the pstn connection. Not necessarily! When you call an analog phone via isdn, the other end will introduce echo so that the ip side will be hearing himself speaking with a small delay. I have that problem with my home BRI running zaphfc. Regards, Tobias Jönsson, Lund SE Hej Tobias, Is this small delay annoying enough? Can it be perceived by the part at the pstn side? Does it disturb fax signals, for example? Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004, Francis Augusto Medeiros wrote: Is this small delay annoying enough? Can it be perceived by the part at the pstn side? Does it disturb fax signals, for example? The echo described by Tobias (originating at the pstn connected user) should only affect the isdn connected part in the scenario above. Whether the echo from the pstn is audioable to the voip user depends on the latency introduced. Fax machines and modems already handle echo and long delays themselves. In fact, an echo canceling or echo supressing device should disable itself when hearing the guard tone (2.1kHz?). It is only annoying to humans. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip to Sip Calls via Asterisk
Hi All, I have a weird problem. I have asterisk setup using the G729 Codec to receive Incoming calls both from a SIP Gateway (SER and Quintum) and via ISDN using i4l and have rules setup in extensions.conf for sending calls out either back via the SIP Gateway or ISDN. What I want to do is have PSTN calls come in via the SIP Gateway, be answered by the auto-attendant and then sent back out to the SIP Gateway to a PSTN number when the particular choice is made. However it gets to the point of ringing and then once the call is connected, there is no voice traffic and the following message appears: chan_sip.c:2752 process_sdp: No compatible codecs! (Only if multiple codecs are available on the SER server) otherwise if I only have one codec allowed on the Quintum and Asterisk, it does not come up with this error (eg G729 or ALAW). However if you ring in from the PSTN (via ISDN) and select this option, it completes the call as requested, the same if I call the menu and select the option from a ip phone connected directly to the Asterisk Box. These Configurations work fine (In Easy step through): Incoming Call from ISDN -- Asterisk Menu -- Selection Made -- Call sent out to SER/Quintum -- Connected Party Incoming Call from local IP Phone -- Asterisk Menu -- Selection Made -- Call sent out to SER/Quintum -- Connected Party Incoming Call from SER from IP Phone -- Asterisk Menu -- Selection Made -- Call sent out to SER/Quintum -- Connected Party This doesn't work: Incoming Call from SER/Quintum from PSTN -- Asterisk Menu -- Selection Made -- Call sent out to SER/Quintum -- Connected Party Everything looks ok here and the configuration is correct (when I can make calls out to the SIP Gateway, both from mentioned earlier and from the IP Phone.) It appears that its only effecting incoming calls coming in from the Quintum from the PSTN to the SER gateway and then to asterisk, which are then being sent back out the SER gateway to the quintum to carry the call back to the PSTN. Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK39591ff8;rport=5060 To: sip:[EMAIL PROTECTED];tag=1bc039ff From: 0388016766 sip:[EMAIL PROTECTED];tag=as34cebb79 Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE Record-Route: sip:[EMAIL PROTECTED];ftag=as34cebb79;lr Contact: sip:[EMAIL PROTECTED]:5061 Content-Type: application/sdp Content-Length: 207 v=0 o=Quintum 13544 2493 IN IP4 192.168.1.90 s=VoipCall c=IN IP4 192.168.1.90 t=0 0 m=audio 10672 RTP/AVP 18 101 c=IN IP4 192.168.1.90 a=rtpmap:18 g729/8000/1 a=rtpmap:101 telephone-event/8000/1 10 headers, 9 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.90:10672 Found description format g729 Found description format telephone-event Capabilities: us - 0x108(ALAW|G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) set_destination: Parsing sip:[EMAIL PROTECTED];lr for address/port to send to set_destination: set destination to 192.168.1.90, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK3fd2c533;rport Route: sip:[EMAIL PROTECTED]:5061 From: 0388016766 sip:[EMAIL PROTECTED];tag=as34cebb79 To: sip:[EMAIL PROTECTED];tag=1bc039ff Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 192.168.1.90:5060 The strange thing is that when this happens it appears that the RTP stream is stable and there is no indication of problems in selecting the codecs. Is there any possible cause as to why this may happen? Especially when it works correctly when I make a call in via the ISDN or an IP phone connected to the Asterisk Server. Does anyone have any pointers as to what may be causing this problem? Thanks, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing Zaptel Modules on Fedora Core 2
On Sat, 14 Aug 2004 15:04:54 -0700, Vikas Deolaliker [EMAIL PROTECTED] wrote: I read a few discussions on installing Zaptel modules in Fedora Core 2 with 2.6.5 kernel. I was wondering if there is a definitive FAQ on this? I am still unable to install by FXO card in my pbx box because the modules won't install. This does work (I have done it with the development kit). The Asterisk Documentation Project is in the process of writing the step-by-step instructions you are looking for. However we are currently lacking on having a few development kits for testing purposes. We are working on getting these, but if people have 2 or 3 they would like to donate, then please contact me off list so we can discuss this futher. Vikas: Sorry for using your message for this, but I have seen this question asked more and more over the last couple of months, so we are in the process of creating what it is you are looking for. However I am currently stuck not writing the Installation chapter due to a lack of resources to purchase the cards. I don't even necessarily need to keep them, I'd be willing to send them back after I was done. Thanks, Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing Zaptel Modules on Fedora Core 2
I have had success with this using both the X100p (wcfxs and wcfxo) and TE410p (wct4xxp) under Redhat FC2 2.6.5. The instructions are on the wiki, do the following: ln -s /lib/modules/2.6.5-1.358/build linux-2.6 cd zaptel make clean make linux26 make install Having said that I have found the TE410p itself to be a bit flakey on our server (HP 1U DL320). The machine will sometimes hardware lock and upon reboot gives a pci bus parity error during post. I have disabled fast post on the server and it hasn't reoccurred but this was only this afternoon so no idea if it has fixed it or not. Craig - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 15, 2004 9:40 PM Subject: Re: [Asterisk-Users] Installing Zaptel Modules on Fedora Core 2 On Sat, 14 Aug 2004 15:04:54 -0700, Vikas Deolaliker [EMAIL PROTECTED] wrote: I read a few discussions on installing Zaptel modules in Fedora Core 2 with 2.6.5 kernel. I was wondering if there is a definitive FAQ on this? I am still unable to install by FXO card in my pbx box because the modules won't install. This does work (I have done it with the development kit). The Asterisk Documentation Project is in the process of writing the step-by-step instructions you are looking for. However we are currently lacking on having a few development kits for testing purposes. We are working on getting these, but if people have 2 or 3 they would like to donate, then please contact me off list so we can discuss this futher. Vikas: Sorry for using your message for this, but I have seen this question asked more and more over the last couple of months, so we are in the process of creating what it is you are looking for. However I am currently stuck not writing the Installation chapter due to a lack of resources to purchase the cards. I don't even necessarily need to keep them, I'd be willing to send them back after I was done. Thanks, Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?
You need a defination for the inbound FWD and what
to do with that.
In my extensions.conf, I have:
[globals]
FWDNUMBER=123456 ;your actual fwd
number
FWDCIDNAME='My Name'
FWDPASSWORD=myfwdpasswd
FWDRINGS=sip/office
FWDVMMBOX=1010
[fwd_out]
exten = _123.,1,SetCallerId,${FWDCIDNAME}
; replace 123 with the desired access code to dial out via FWD
exten =
_123.,2,Dail(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN}:3},60,r)
exten = _123.,3,Congestion
[local]
include = fwd_out :add to local
context
[default]
;inbound dialing from FWD
exten =
${FWDNUMBER},1,Goto(housemenu,s,1) ; I have mine set to hit a menu, no
reason you cann't forward to an extension instead
- Original Message -
From:
Chris Blunt
To: [EMAIL PROTECTED]
Sent: Sunday, August 15, 2004 3:29
AM
Subject: [Asterisk-Users] Inbound Free
World Dialup - extension not ringing?
Hi to all the * people out
there,
Please kind to me as I am both new
to Asterisk and to Linux But I am learning
fast.
My config is quite simple, Im
just following examples and the Wiki: I have two PCs running X-Lite
phones, these work without problems between each other, and I have a GS
BudgeTone-100 registered to Free World Dial UP (working no
problem).
I have tried to set up Asterisk to
accept calls from FWD on another number I have registered, but I cant get my
local X-Lite to ring on an inbound call from FWD, and I get the busy tone on the
BT100
When I sip debug, I can see that I
am registered with FWD, and when I call the number from the BT100 I can see
all the incoming information but still nothing on my
X-Lite.
My
extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
[sip]
exten =
1,1,Dial(SIP/phone1,20,tr)
exten =
2,1,Dial(SIP/phone2,20,tr)
exten =
2,2,VoiceMail,u1234
exten =
2,102,VoiceMail,b1234
;exten =
1000,1,Dial(SIP/phone1SIP/phone2,20,tr)
exten =
1001,1,Ringing
exten =
1001,2,Wait(2)
exten =
1001,3,VoicemailMain,s1234
exten =
6601,1,WaitMusicOnHold(60)
exten =
232999,1,Dial(SIP/phone1,30,tr)
exten =
232999,2,Hangup
I am behind a NATed fire wall, but
Im not sure that is related.
Any ideas or help (working simple
confs) would be much appreciated.
Best
regards
--
Chris
Blunt
SIP:
[EMAIL PROTECTED]
Re: [Asterisk-Users] chan_oh323 loading error
You might try setting P_PTHREADS=1 in your Makefile. I'm not actually certain if this will work, but it can't hurt anything. Ryan Wilkins On Sun, 15 Aug 2004, Krystian Filiks wrote: I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what that mean? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 loading error
Hi all! Any one that could give me some input on the problem below? regards Krystian Krystian Filiks wrote: I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what that mean? Regards Krystian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 loading error
Would the command make P_PTHREADS=1 opt do the job? Krystian Ryan Wilkins wrote: You might try setting P_PTHREADS=1 in your Makefile. I'm not actually certain if this will work, but it can't hurt anything. Ryan Wilkins On Sun, 15 Aug 2004, Krystian Filiks wrote: I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what that mean? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CCM -(H323) - *
My hack worked for me, and still does and last time I checked was still needed. There's no warranty for anyone else. It's possibile there's a cleaner way to fix it, but I've not found it. It was a one line addition to the OpenH323 library source that chan_h323 links against - you don't modify Asterisk itself at all. Read the original message in the thread you posted a link to, it explains what I saw and what I did about it. FWIW, it was CCM 3.something. Chris. Andres Junge wrote (on Aug 12): Hi I have found in http://lists.digium.com/pipermail/asterisk-users/2004-July/056111.html (Hack to make * - (H323) - CCM - IOS GW work) that i need a special version of chan_h323, because of the External RTP problem. Do you know exactly which version is it? Or do i need an unofficial patch? Thanx Andr?s ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto remove digits from a called number
Greg Hill a écrit :
On Sat, 14 Aug 2004, administrator tootai wrote:
Hi list,
I have SIP clients and H323 GK connected through h323 channel (Nufone).
In h323 conf I gave prefix=09 so all call starting with this prefix are
send to asterisk. The context is also given their as [fromh323]
But now, in asterisk, I want to have the called number without this 2
leading digits so the exten variable will be according to my actual
dialplan. Here's an exemple:
In extensions.conf I have
exten = 100,1,Goto(demo,s,1)
If I call #100 from SIP it's ok. So now, if I want to reach this
extension from an h323 EP, I have to call 09100. This call will never
succeed (or I create a new exten line, same as above, with this prefix).
You're right, you will have to create an extension to match the 09xxx
numbers. But you don't have to create one for every real SIP extension
you have. Instead, make one that matches all 09xxx extensions and does a
goto:
exten = _09XXX,1,Goto(yoursipextensionscontext,${EXTEN:2},1)
That did the trick.
for three digit real extensions. Add or remove X's for more or fewer
digits, or just use _09. for _ANYTHING_ that starts with 09 (keep that in
mind.. sometimes that wildcard extension comes back to bite you!).
Already done ;-) Thanks for you help
--
Daniel
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Re: [Asterisk-Users] chan_oh323 loading error
No. Look in the Makefile of the oh323 driver source. Search down through the file for #P_PTHREADS=1 and remove the #. Then recompile. See if that helps your situation any. It may.. or it may not. On Sun, 15 Aug 2004, Krystian Filiks wrote: Would the command make P_PTHREADS=1 opt do the job? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free MOH MP3
CVS has them - Original Message - From: Wiley E. Siler [EMAIL PROTECTED] Date: Sat, 14 Aug 2004 16:50:43 -0700 Subject: [Asterisk-Users] Free MOH MP3 To: [EMAIL PROTECTED] Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone's post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Hi Francis, Francis Augusto Medeiros wrote: Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and standards as a PBX-only, with traditional phones? Or should I go all the way to Digium's TDM? Or should I forget the whole thing and get a traditional PBX? ;) If you already have the analog telephone wiring in place, and you are on a budget, I recomend you to use sipura spa-2000 adapters. They are a whole lot better than GS phones. You can have 3way conferences and attendant transfers. With GS you cannot do that. The price is as good for the sipuras as the GS phones, about $50 per FXS port, plus a cheap analog phone and you will be all set. My concerns are most latencies. Our network will be a switch with lots of ports, all 100mb/s, with VERY low traffic. Internal calls (SIP to SIP) will sound great. You will probably experience some echo when going to POTS. I did not try the Sipura SPA-3000 yet, but it seems to be a cheap alternative to a gateway, providing you with one FXO and one FXS for $130 or so. the echo cancellation in the sipura works well for fxs, it might work well to for fxo. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vlan question
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: 13 Aug 04, 2:38 PM Subject: [Asterisk-Users] Vlan question Hi, I am setting up an Asterisk system with Cisco 7960 phones. I have a PoE injector to insert between the patch panel and HP 2626 switch. I plan to plug the users pc into the phone and the phone into the wall. I would like the phones to have a seperate subnet from the phones for performance reasons. May be a silly question, but with the pc and phone sharing the same switch port, how will it know to seperate the traffic and subnets? Thanks Hmm, virtual IP on the NIC would be the way. I've not done anything virtual for so many years so I could be wrong. But you will not save performance when you share the same wire. The switch learns who is hosting what IP so that's not the problem. It should also be able to route multiple LANs. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBH4m/ljK16xgETzkRAocPAJ4q5QGR3Mo0RXO+hzQ0+cTTpCk4sACfZxSp ZD+mM2CHEEKUjcKUs+m2Y9Y= =CheB -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?
From ethereal traces, I noticed that the HT-286 will only send out DTMF signals if it's the calling party, so the act of flashing somehow makes it think that it is now the calling party, so DTMF signals were sent. Not sure if that is a bug or a feature with a certain design consideration. I discovered that method quite by accident, really, since I can't find any documentation about that method in the user manual. - Original Message - From: Dennis Cartier [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 15, 2004 7:51 PM Subject: Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm? Thanks!!! That works. A bit cumbersome but at least it works. Why does putting someone on hold and then taking them off again make the ATA (or *) recognize the # key as transfer? Thankyou for your help On Sun, 15 Aug 2004 11:49:50 +0800, MPlus [EMAIL PROTECTED] wrote: For blind transfers, press flash twice, then press #. For consultative transfers, press flash once, talk to the other party and tells him to hangup, press flash again, then press #. - Original Message - From: Dennis Cartier [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 15, 2004 2:10 AM Subject: Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm? Unfortunately the Cisco ATA-186 does not support iLBC which means extra costs for purchasing 729 licenses. The ATA-286 works fine other than this 1 issue. Do the Grandstream developers follow this list?? This problem has been persistent for a LONG time and each new firmware version still has it unfixed!! On Sat, 14 Aug 2004 19:36:43 +0300, Lubomir Christov [EMAIL PROTECTED] wrote: Yes, we are experiencing the same problem and because of that we switched the called HT ata to Cisco ATA 186 ... Lubo Andy Lee wrote: [ Message pasted from Sun, 7 Mar 2004 08:27:54 -0600 ] The problem I'm experiencing with many GS adapters, regardless of firmware version is this. Call from one phone to another phone using both the 'T' and 't' flags in the Dial() command. After they are connected, you should be able to press '#' on either phone to hear transfer. What I am experiencing is the calling GS adapter will hear transfer when they press '#', but when the receiving GS adapter presses '#', nothing happens at all. Are you able to repeat this? If not, can you please tell me the firmware revisions and Asterisk version that you are using? Thank you very much. Yes, I am experiencing the same problem. Works fine with the BudgetTone phones but not with the HandyTone-286. Have you resolved this yet? We've been trying to figure this out on and off for the past couple of weeks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Appradius Project: RADIUS authentication and accounting support for Asterisk PBX http://appradius.minitelecom.org/ - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free MOH MP3
Wiley E. Siler [EMAIL PROTECTED] wrote: [...] Does anyone know where I can get some royalty free, cost free music for my music on hold? The stuff at www.zongoftheweek.com is CC-licensed so should be fair game. Whether you want to inflict some of it on callers is another matter :) -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 loading error
I have searched through the Makefile and the configure files and could not find any instance of P_PTHREADS. Should I put it there? in that case where? Ryan Wilkins wrote: No. Look in the Makefile of the oh323 driver source. Search down through the file for #P_PTHREADS=1 and remove the #. Then recompile. See if that helps your situation any. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 13:03:49 -0300, Nicolas Gudino [EMAIL PROTECTED] wrote: Hi Francis, If you already have the analog telephone wiring in place, and you are on a budget, I recomend you to use sipura spa-2000 adapters. They are a whole lot better than GS phones. You can have 3way conferences and attendant transfers. With GS you cannot do that. The price is as good for the sipuras as the GS phones, about $50 per FXS port, plus a cheap analog phone and you will be all set. The thing is that it's a new office, so we can choose what kind of wiring to use... My concerns are most latencies. Our network will be a switch with lots of ports, all 100mb/s, with VERY low traffic. Internal calls (SIP to SIP) will sound great. You will probably experience some echo when going to POTS. I did not try the Sipura SPA-3000 yet, but it seems to be a cheap alternative to a gateway, providing you with one FXO and one FXS for $130 or so. the echo cancellation in the sipura works well for fxs, it might work well to for fxo. Gracias Nicolas! I'll really give it a look... Too bad that with this option I'll loose the LCD's, but, what the heck... ;) Cheers, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 loading error
Out of my league.. It may.. You can always try it. On Sun, 15 Aug 2004, Krystian Filiks wrote: I think that I found it, I'm compiling PWLIB with ./configure --with-pthreads Do you think this would do it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sunday 15 August 2004 12:03, Nicolas Gudino wrote: If you already have the analog telephone wiring in place, and you are on a budget, I recomend you to use sipura spa-2000 adapters. They are a whole lot better than GS phones. You can have 3way conferences and attendant transfers. With GS you cannot do that. The price is as good for the sipuras as the GS phones, about $50 per FXS port, plus a cheap analog phone and you will be all set. Why on earth would you install SPA-2000s and endure that wiring mess? An FXS channel bank and a BIX strip will save you YEARS in lost time due to wiring and general messiness! -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP connections do not hang up
Hello all,
I also have this SIP CANCEL problem and have inverstigated the problem a
bit but am not sure if the problem lies in the sipgate proxy or asterisk:
1.) This only happens when you CANCEL an INVITE (obviously) the INVITE
is shown below.
2.) sipgate sends a 183 Session Prgress response message to asterisk
shown below as an example.
3.) To hand up Asterisk sends a CANCEL message to sipgate and it is
happy and acknowledges.
4.) The called parties phone contiunes to ring (incorrectly)!
5.) The problem happens because the CANCEL SIP line contains a different
contact to the original INVITE massage which according to the SIP RFC
3261 is illegal.
from RFC 3261 chapter 9.1
The Request-URI, Call-ID, To, the numeric part of CSeq, and From header
fields in the CANCEL request MUST be identical to those in the
request being cancelled, including tags.
I assume the Request-URI in this case is defined as
sip:[EMAIL PROTECTED] in my exaple INVITE message. The CANCEL
contains the URI which can be found in the 183 Session Progress message
set to be sip:[EMAIL PROTECTED]
Asterisk chan_sip has the following code part that handles all other
SIP/2.0 messages (e.g. not specifically processed) in function
handle_request():
} else if (!strcasecmp(cmd, SIP/2.0)) {
extract_uri(p, req);
The function extract_uri updates the URI of the current context, as this
has changed in the 183 Session Progress then the URI used in the CANCEL
message also changes.
My conclusion is the Asterisk probably handles this wrongly as it does
not happen with soft phones (they use the original INVITE URI). I have
curently commented out the code part and it fixes the problem I have
described BUT I do not know enough about the code to say why it was
included in the first place! So it would be nice to submit this problem
to the developer to get his answer and let us know what he/she thinks
should happen. Does anyone know how to submit this kind of request?
Let me know your opinions.
Ian Hailey.
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.187.52.208:5060;branch=z9hG4bK748c9b3f
From: 5337478 sip:[EMAIL PROTECTED];tag=as31d0bbe0
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 13 Aug 2004 05:50:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 243
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 217.187.52.208:5060;branch=z9hG4bK748c9b3f
From: 5337478 sip:[EMAIL PROTECTED];tag=as31d0bbe0
To: sip:[EMAIL PROTECTED];tag=as0deea40d
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
Content-Length: 240
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.187.52.208:5060;branch=z9hG4bK748c9b3f
From: 5337478 sip:[EMAIL PROTECTED];tag=as31d0bbe0
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
Florian Rau wrote:
Hi,
Well, the Problem is not the ZAP Channel but the SIP Channel, because it
occurs no matter what channel I use the phone outside. Maybe this graph is
more descriptive:
1. ZAP or SIP == 2. Asterisk == 3. SIP (thru internet, provider sipgate)
== 4. PSTN
The connections on 1. hang up correctly, as seen in the log, but the SIP
connection of 3. does NOT hangup.
Regards,
Florian
PS: Believe me, I'm searching for over one week in the whole internet for a
solution, but did not find it.
- Original Message -
From: Jean-Yves Avenard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 30, 2004 11:46 PM
Subject: Re: [Asterisk-Users] SIP connections do not hang up
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
If you just bothered to search this list in the past 12 hours, you
would have found a solution around that:
to summarize:
Add in zapata.conf:
busydetect=yes
busycount=6
The maximum it will take for asterisk to see the person hanged-up is
after 6 busy dial-tones.
On 31/07/2004, at 6:58 AM, Florian Rau wrote:
I'm calling from inside (either X-Lite using SIP channel or a ISDN
telephone
using Zap Channel) using sipgate to a number in public network.
When I'm hanging up before the other person picked up the phone, the
line is
not closed correctly.
The phone keeps on ringing until timeout (of Sipgate I assume) and it
even
costs my money, if the other person picks up the ringing phone, even
if I
already hung up.
- ---
Jean-Yves Avenard
Hydrix Pty Ltd - Embedding the net
www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (Darwin)
iD8DBQFBCsHLXeDVKqIr3GURArjyAJ9p97F/wWIiIesaYo85QfHut8zbzQCgj2l2
uuKZxyJoaSmpI9V9I+ojnJc=
=Y8jQ
-END PGP SIGNATURE-
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[Asterisk-Users] GrandStream ATA286 RC2 (was RC2 - H323 channel broken)
Hello everybody, when I upgraded from RC1 to RC2 I didn't had any audio between my ATA286 and H323 EP (my post from 13/08/04) I checked further and discover that problem is with ATA286 who is unable to call. I always get an 404 error. Coming back to RC1 everything works fine again. I tried to modify my dtmfmode from rfc2833 to info but in change nothing. Local call to asterisk are working (playing music or voicemail box) and, as said above, calling an h323 EP is ringing but no audio. Thanks for any hint. -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 13:39:10 -0400, Andrew Kohlsmith [EMAIL PROTECTED] wrote: Why on earth would you install SPA-2000s and endure that wiring mess? An FXS channel bank and a BIX strip will save you YEARS in lost time due to wiring and general messiness! Hello Andrew! I'm sorry to ask this really, reeally newbie thing, but... what would be an FXS channel bank, and where would I find more info about some popular models? And the same question goes to... BIX strips! What are those?? :) Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 loading error
Ryan Wilkins wrote: Maybe you are not running the latest version of oh323.. I'm running 'asterisk-oh323-0.6.3a'. That is what I'm trying to get going as well with openH323 1.13.5 and pwlib 1.6.6 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk MIBS
Title: Asterisk MIBS Hi, I was wondering if there are any Asterisk MIBS (specifically regarding call information) ? I noticed a post citing www.faino.org, but this site doesnt seem to exist anymore, and The Book v2 doesnt have any references to MIBS. Any pointers greatly appreciated. Keith Burns The dogs may bark but the caravan rolls on
RE: [Asterisk-Users] Free MOH MP3
William Suffill [EMAIL PROTECTED] top-posted: CVS has them That hasn't been established yet, to my knowledge. The music in CVS appears to have come from a source that doesn't allow free commercial use (www.freeplaymusic.com, according to the CREDITS file). Music on hold is classed as commercial use. Correct me if I'm wrong, but this point has been raised before and I haven't seen a clarification. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sunday 15 August 2004 13:50, Francis Augusto Medeiros wrote: I'm sorry to ask this really, reeally newbie thing, but... what would be an FXS channel bank, and where would I find more info about some popular models? And the same question goes to... BIX strips! What are those?? :) haha A channel bank just serves as a device to convert channelized T1s into phone lines and back. FXS ports let you plug phones into them, FXO ports you plug phone lines in to. A BIX strip is just a common wiring closet item that lets you terminate 25 pairs to a strip about 5 or 6 inches long. Google Image search will show you exactly what they look like. A channel bank would typically terminate to a D50F connection, which you would then use a D50M to BIX cable -- this lets you easily terminate 24 phones to a T1, VASTLY reducing the mess and wiring hassle. Since you're moving in to a new place and you don't have existing wiring to make use of I'm not sure any of this would be of any use to you. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free MOH MP3
That could right don't really use MOH much but I noticed there was in CVS. Although why would it be in CVS of asterisk if not used for MOH though? On Sun, 15 Aug 2004 18:57:39 +0100, Kevin Walsh [EMAIL PROTECTED] wrote: William Suffill [EMAIL PROTECTED] top-posted: CVS has them That hasn't been established yet, to my knowledge. The music in CVS appears to have come from a source that doesn't allow free commercial use (www.freeplaymusic.com, according to the CREDITS file). Music on hold is classed as commercial use. Correct me if I'm wrong, but this point has been raised before and I haven't seen a clarification. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk with InPhonex?
hello has any one got asterisk to work with InPhonex? if so can you send me your conf information? we are having some problems getting ours up and running. my friend is helping me get it set up. thanks hank My Inbox is protected by SPAMfighter415 spam mails have been blocked so far.Download free SPAMfighter today!
RE: [Asterisk-Users] Free MOH MP3
William Suffill [EMAIL PROTECTED] lazily top-posted: That could right don't really use MOH much but I noticed there was in CVS. Although why would it be in CVS of asterisk if not used for MOH though? That's the part that needs clarification. Perhaps the music has been included to help people test their MoH setup and no commercial use license has been granted. In this case, the music should not be included as part of any GPLed package, and the presence of the files in the Asterisk CVS archive would be questionable; It would be better to include a comment in the musiconhold.conf file to show people where test music files can be found. On the other hand, perhaps the people at Freeplaymusic have granted all Asterisk users a license to use these specific tracks for MoH, and have given their consent for their music to be included as part of Asterisk. It's also possible that someone at Freeplaymusic has signed one of the Digium disclaimers or has released those specific music files under the GPL, the CCL or a similarly free license. The point is that, unless I've missed something, no clarification has been posted. If no specific permission has been granted then the published Freeplaymusic license remains in force. This means that commercial use, including use as music on hold, is forbidden. I would advise against using the supplied music as MoH until the permission to do so has been shown. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Free MOH MP3
In article [EMAIL PROTECTED], Kevin Walsh [EMAIL PROTECTED] wrote: I would advise against using the supplied music as MoH until the permission to do so has been shown. I've just submitted bug #2255 concerning this very point, as that seems to be the best way to get action on something. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:Re:7960 help
- Original Message - Message: 2 Date: Sat, 14 Aug 2004 20:48:55 -0700 (PDT) From: Gonzalo Gasca Meza [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7960 help To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] --0-1412799478-1092541735=:62359 Content-Type: text/plain; charset=us-ascii hi man, if you are trying to upgrade to the latest version, change the permissions of the file, then to the SIPmacaddress.cnf file add a line that says image version = version, copy that line from the Sipdefault.cnf file, . If the first workaround does not work, try to downgrade to version 2.3 and the do the upgrade directly from that version. I can provide you any image you need. Let me know how that works I will highly appreciate your answer ok, so I tried the first suggestion by adding 'image_version= POS3-06-3-00 to the beginning of the SIPMAC.cnf file for one of the phones that is having trouble but still no luck. I do not have access to any other images so if you want to send me one on or off list that would be great. Jason Kawakami Jason Kawakami [EMAIL PROTECTED] wrote: I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] __use_ast_pthread_create_instead__
Please anyone, When I start * after installing the asterisk-oh323-0.6.3a I get [chan_oh323.so]Aug 15 22:36:44 WARNING[1076252800]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 22:36:44 WARNING[1076252800]: loader.c:423 load_modules: Loading module chan_oh323.so failed! I followed the instructions to the . What am I doung wrong? I tried recompiling everything and still no joy! Thanks Krystian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Internal Distinctive Ringing + Caller ID
I have set up my asterisk PBX to provide a double-ring for outside calls, and a single ring for station-to-station. (I'm talking about ZAP stations in this email). I had to go into one of the .c files and tell it to expect the Caller ID between the 2nd and 3rd rings in order to get the double-ring scenario to work. My problem is that, in making this change, I now don't see Caller ID on internal calls. Is there a work-around for this? It'd be really handy to have caller ID on both internal and external calls, AND to continue to have the distinctive ringing that I've been using. (I use the 'r5' option in my Dial(ZAP/) statement to get the double ring). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream ATA286 RC2 (was RC2 - H323 channel broken)
administrator tootai a écrit : Hello everybody, when I upgraded from RC1 to RC2 I didn't had any audio between my ATA286 and H323 EP (my post from 13/08/04) I checked further and discover that problem is with ATA286 who is unable to call. I always get an 404 error. Coming back to RC1 everything works fine again. I tried to modify my dtmfmode from rfc2833 to info but in change nothing. Local call to asterisk are working (playing music or voicemail box) and, as said above, calling an h323 EP is ringing but no audio. Thanks for any hint. Seems to be really an h323 issue: 404 disappear (config error) but still can't connect to h323 EP: calling from an H323 EP to ATA is ok, but from ATA to H323 EP no audio on both sides. Coming back to RC1 it's just running fine. In h323.conf I allow g711u, g711a and gsm. Call from h323 EP to ATA is done in gsm, call from ATA to H323 is g711. Is there something broken with RC2? -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MIBS
Alagalah wrote: Hi, I was wondering if there are any Asterisk MIBS (specifically regarding call information) ? I noticed a post citing ___www.faino.org_ http://www.faino.org, but this site doesnt seem to exist anymore, and The Book v2 doesnt have any references to MIBS. Any pointers greatly appreciated. Check here for Asterisk and SNMP support: http://faino.it/en/ast-ax-snmpd.html /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no tones detected
that just clicked thank you the sample app they included had some freqs set to listen for keypresses I wonder if something is off over there, must be voicetronix's fault On Sun, 15 Aug 2004 15:44:03 -0600 (MDT), Greg Hill [EMAIL PROTECTED] wrote: On Sun, 15 Aug 2004, Johnathan Bunn wrote: maybe this has been covered before but, i can't find it, has anyone had a problem where outside lines can't use number presses like choose extensions but inside lines can, I am using voicetronix hardware with asterisk and when i call from a station port I hear my greeting and can dial an extension and connect, but if I call in I can here my greeting and pushing buttons does nothing, and they dont show up on the console either I have tried 3 land line phones... any help or a point in the right direction would be extremely helpful I don't know anything about your hardware, but it sounds like yours is a DTMF mode problem. There are multiple ways of communicating the DTMF tones in the land of VOIP. SIP, for example, typically uses RFC2833, info, or inband. You set up * (or it defaults) to expect DTMF in a particular format, and it ignores the other formats. So if the far end transmits inband and asterisk is expecting RFC2833 (default on SIP connections) the it won't recognize the caller's button presses. Search google and the wiki for dtmfmode or dtmf mode for your device's connection method (sip/zap/iax/whatever). Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- To follow the Path, Look to the master, Walk with the master, See through the master, Become the master ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Teliax TOS copied from Vonage?
TelIAX, one of the new VoIP-to-PSTN gateway providers, has their terms of service posted on their signup page: http://teliax.com/user_admin/signup/s1.php They look strangely familiar--it's exactly the same as http://www.vonage.com/features_terms_service.php with s/Vonage/Teliax/. (And it's cut off about halfway through). Anyone else notice this? Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?
IAX2 uses udp port 4569, so youll probably have to open that up on your
firewall/router.
http://www.voip-info.org/ is a good starting place for any asterisk problems
- specifically:
http://www.voip-info.org/wiki-Asterisk+firewall+rules
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
HTH
Ed
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Blunt
Sent: 15 August 2004 23:06
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?
Hi Lyle,
Thank you so much for your help, I think your information points to using
IAX2 rather than registering with FWD from the sip.conf
I have made an attempt to understand this, added the appropriate information
into iax.conf, remove old info from sip.conf, gone to fwd and ticked the IAX
registration box, and I now get my local sip phone ringing when I dial in
from FWD! Hurrah, unfortunately I get no sound in either direction. Do
you have any experience of this or could it be due to me being inside a NAT
firewall? I have port 5060 forwarded to my * server, should I forward any
other ports? (I can only forward a maximum 20 single ports due to a
limitation on my home router).
As yet I am unable to make outgoing calls over FWD, I figured I would look
at this next.
Is there a NAT solution that could be used with sip.conf rather than the
IAX?
Again your help is most appreciated.
Best regards
Chris
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: 15 August 2004 15:14
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Inbound Free World Dialup - extension not
ringing?
You need a defination for the inbound FWD and what to do with that.
In my extensions.conf, I have:
[globals]
FWDNUMBER=123456 ; your actual fwd number
FWDCIDNAME='My Name'
FWDPASSWORD=myfwdpasswd
FWDRINGS=sip/office
FWDVMMBOX=1010
[fwd_out]
exten = _123.,1,SetCallerId,${FWDCIDNAME} ; replace 123 with the desired
access code to dial out via FWD
exten =
_123.,2,Dail(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN}:3},60
,r)
exten = _123.,3,Congestion
[local]
include = fwd_out :add to local context
[default]
;inbound dialing from FWD
exten = ${FWDNUMBER},1,Goto(housemenu,s,1) ; I have mine set to hit a
menu, no reason you cann't forward to an extension instead
- Original Message -
From: Chris Blunt
To: [EMAIL PROTECTED]
Sent: Sunday, August 15, 2004 3:29 AM
Subject: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?
Hi to all the * people out there,
Please kind to me as I am both new to Asterisk and to Linux But I am
learning fast.
My config is quite simple, Im just following examples and the Wiki: I have
two PCs running X-Lite phones, these work without problems between each
other, and I have a GS BudgeTone-100 registered to Free World Dial UP
(working no problem).
I have tried to set up Asterisk to accept calls from FWD on another number I
have registered, but I cant get my local X-Lite to ring on an inbound call
from FWD, and I get the busy tone on the BT100
When I sip debug, I can see that I am registered with FWD, and when I call
the number from the BT100 I can see all the incoming information but still
nothing on my X-Lite.
My extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
[sip]
exten = 1,1,Dial(SIP/phone1,20,tr)
exten = 2,1,Dial(SIP/phone2,20,tr)
exten = 2,2,VoiceMail,u1234
exten = 2,102,VoiceMail,b1234
;exten = 1000,1,Dial(SIP/phone1SIP/phone2,20,tr)
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain,s1234
exten = 6601,1,WaitMusicOnHold(60)
exten = 232999,1,Dial(SIP/phone1,30,tr)
exten = 232999,2,Hangup
I am behind a NATed fire wall, but Im not sure that is related.
Any ideas or help (working simple confs) would be much appreciated.
Best regards
--
Chris Blunt
SIP: [EMAIL PROTECTED]
___
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Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?
Thanks for the workaround! Hmm...I think it is an unwanted feature. I hope the GrandStream developers fix this soon. The workaround isn't something that end-users are going to remember easily or like. On Mon, 16 Aug 2004 00:08:40 +0800, MPlus [EMAIL PROTECTED] wrote: From ethereal traces, I noticed that the HT-286 will only send out DTMF signals if it's the calling party, so the act of flashing somehow makes it think that it is now the calling party, so DTMF signals were sent. Not sure if that is a bug or a feature with a certain design consideration. I discovered that method quite by accident, really, since I can't find any documentation about that method in the user manual. - Original Message - From: Dennis Cartier [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 15, 2004 7:51 PM Subject: Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm? Thanks!!! That works. A bit cumbersome but at least it works. Why does putting someone on hold and then taking them off again make the ATA (or *) recognize the # key as transfer? Thankyou for your help On Sun, 15 Aug 2004 11:49:50 +0800, MPlus [EMAIL PROTECTED] wrote: For blind transfers, press flash twice, then press #. For consultative transfers, press flash once, talk to the other party and tells him to hangup, press flash again, then press #. - Original Message - From: Dennis Cartier [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 15, 2004 2:10 AM Subject: Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm? Unfortunately the Cisco ATA-186 does not support iLBC which means extra costs for purchasing 729 licenses. The ATA-286 works fine other than this 1 issue. Do the Grandstream developers follow this list?? This problem has been persistent for a LONG time and each new firmware version still has it unfixed!! On Sat, 14 Aug 2004 19:36:43 +0300, Lubomir Christov [EMAIL PROTECTED] wrote: Yes, we are experiencing the same problem and because of that we switched the called HT ata to Cisco ATA 186 ... Lubo Andy Lee wrote: [ Message pasted from Sun, 7 Mar 2004 08:27:54 -0600 ] The problem I'm experiencing with many GS adapters, regardless of firmware version is this. Call from one phone to another phone using both the 'T' and 't' flags in the Dial() command. After they are connected, you should be able to press '#' on either phone to hear transfer. What I am experiencing is the calling GS adapter will hear transfer when they press '#', but when the receiving GS adapter presses '#', nothing happens at all. Are you able to repeat this? If not, can you please tell me the firmware revisions and Asterisk version that you are using? Thank you very much. Yes, I am experiencing the same problem. Works fine with the BudgetTone phones but not with the HandyTone-286. Have you resolved this yet? We've been trying to figure this out on and off for the past couple of weeks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Appradius Project: RADIUS authentication and accounting support for Asterisk PBX http://appradius.minitelecom.org/ - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or
[Asterisk-Users] Newbie with missing .conf files
Yeah, I'm a newbie and am having problems with missing .conf files in /etc/asterisk/ I get notices when I try to run asterisk like: parking.conf is depreciated in favour of features.conf Please rename it. (I'm not getting rename parking .conf to features.conf ???) I am also missing: mgcp.conf enum.conf cdr.conf I using dev-lite Digium board that has 1 fxo and 1fxs port. So far I was able to load and configure the files unfortunately I can not setup my voice mail message it says I missing some hooks??? How can I tell which version of asterisk I'm using and can anyone help me with the above problems? Oh, If I'm not detail enough, please advice me? Sincerely, Don Moskaluk [EMAIL PROTECTED] www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home
[Asterisk-Users] 123 Basic configuration files
I need to find some basic configuration files. Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports? Don Moskaluk [EMAIL PROTECTED] www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home
RE: [Asterisk-Users] 123 Basic configuration files
Best starter examples http://www.automated.it/guidetoasterisk.htm Documentation http://www.digium.com/index.php?menu=documentation Asterisk will make sample files for you... read teh doucmentation at the first link I listed... Regards, Wiley I need to find some basic configuration files. Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports? Don Moskaluk [EMAIL PROTECTED] www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 13:58:58 -0400, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Sunday 15 August 2004 13:50, Francis Augusto Medeiros wrote: I'm sorry to ask this really, reeally newbie thing, but... what would be an FXS channel bank, and where would I find more info about some popular models? And the same question goes to... BIX strips! What are those?? :) haha A channel bank just serves as a device to convert channelized T1s into phone lines and back. FXS ports let you plug phones into them, FXO ports you plug phone lines in to. A BIX strip is just a common wiring closet item that lets you terminate 25 pairs to a strip about 5 or 6 inches long. Google Image search will show you exactly what they look like. A channel bank would typically terminate to a D50F connection, which you would then use a D50M to BIX cable -- this lets you easily terminate 24 phones to a T1, VASTLY reducing the mess and wiring hassle. Since you're moving in to a new place and you don't have existing wiring to make use of I'm not sure any of this would be of any use to you. Thanks Andrew! Well, those options are kinda way above my needs, as we won't have that many phone lines nor extensions. Cheers, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MIBS
Alagalah wrote: Hi, I was wondering if there are any Asterisk MIBS (specifically regarding call information) ? I noticed a post citing www.faino.org http://www.faino.org , but this site doesn t seem to exist anymore , and The Book v2 doesn t have any references to MIBS. I compiled this into RC1, and all I got was congestion tones when I tried to use *. Also, it requires ucd-snmp, the older version of net-snmp. I'm going to look at Nagios for remote monitoring. Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Andrew Kohlsmith wrote: On Sunday 15 August 2004 12:03, Nicolas Gudino wrote: If you already have the analog telephone wiring in place, and you are on a budget, I recomend you to use sipura spa-2000 adapters. They are a whole lot better than GS phones. You can have 3way conferences and attendant transfers. With GS you cannot do that. The price is as good for the sipuras as the GS phones, about $50 per FXS port, plus a cheap analog phone and you will be all set. Why on earth would you install SPA-2000s and endure that wiring mess? An FXS channel bank and a BIX strip will save you YEARS in lost time due to wiring and general messiness! I prefer the wiring mess and sipuras than the GS phones. That's all. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modified Prepaid doesn't update the balance
Hi all, Has anyone successfully patch the modified prepaid application to update the balance on the card after a call? Best regards, Glynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
