[asterisk-users] Asterisk both behind a NAT and outside at the same time
I've read a lot of the descriptions of handling NAT with Asterisk, and the use of both the nat and canreinvite flags. I am very familiar with Sip and NAT but have not seen an answer to the following question. My Asterisk server runs on a machine with two ethernets. One is an external net, with exposed IP addresses. The other is an internal net with natted IP addresses. Thus the server has two addresses. The server is _not_ the NAT gateway. That's a linksys box which has its own external IP to gateway traffic from the internal natwork. The phones are on the internal NATwork. Asterisk talks to them over it. Outside peers, such as SIP termination providers etc. talk to the Asterisk server via its outside address, which is as you would expect. However, from time to time I get the famous one-way audio because Asterisk has decided to do a native bridge between a natted SIP phone and an external SIP peer. It sends the internal IP of the SIP phone in the SDP and of course the outside service can't send packets to that. I could just turn off reinvites on the internal phones, but this would cause them to route all traffic through the asterisk box, even on internal calls between phones on the same ethernet, which seems foolish to me. I don't want to turn off reinvites to the external peers -- if a call comes in from a SIP originator for example, and is send back out to a SIP terminator (call forwarding) I want a native bridge for sure.(Handling the internal traffic is not so much of a burden though sometimes I hear latency because of it, but routing external traffic through the asterisk box is a bad thing.) So what I want is for Asterisk to use native bridges when connecting two channels behind the NAT, or two channels on the real internet, but not to do so when connecting an internal and external channel. It should be able to see the IP addresses, and know the difference between natted and external ones and know they can't talk to one another. (The ICE protocol would handle this someday.) Is IAX smarter about this? Of course I might even want to get smarter about this. Is it possible, typically by configuring stun in the phones, to have them be aware of their external IP and tell Asterisk about it? With a full cone NAT, it would work to do a native bridge between the internal and external devices so long as the external device is given the right address and port of the NAT box, not the internal address of the phone. However, we don't want to do this on internal to internal calls -- many NATs can't hairpin. I would think this would be a common situation (though perhaps more commonly the asterisk server IS the firewall/NAT.) Is there a solution that does the right thing most of the time? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fedora Core 6 (FC6) and Asterisk-1.2.13 and Zaptel-1.2.10 compile problems
All, I have upgraded by home machine from Fedora Core 5 (FC5) to the recent FC6 and am struggling to build Zaptel-1.2.10 and Asterisk-1.2.13 on the box... which is an Intep P4 2.8GHz HT processor box with 845 chipset, hence the kernel installed is 2.6.18-1.2798.fc6-i686so we hve this: [EMAIL PROTECTED] zaptel-1.2.10]# uname -a Linux gate.tubby.org 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:54:20 EDT 2006 i686 i686 i386 GNU/Linux I have ensured that the kernel-devel package and glibc-kernel headers are installed. When I attempt to compile Zaptel-1.2.10 I get the following: make[1]: Entering directory `/usr/src/kernels/2.6.18-1.2798.fc6-i686' CC [M] /root/asterisk/zaptel-1.2.10/zaptel.o In file included from /root/asterisk/zaptel-1.2.10/zaptel.c:40: /root/asterisk/zaptel-1.2.10/zconfig.h:9:26: error: linux/config.h: No such file or directory make[2]: *** [/root/asterisk/zaptel-1.2.10/zaptel.o] Error 1 make[1]: *** [_module_/root/asterisk/zaptel-1.2.10] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-1.2798.fc6-i686' make: *** [linux26] Error 2 and can confirm that the file does not exist in the path to the kernel source: /usr/src/kernels/2.6.18-1.2798.fc6-i686/include/linux/ Grabbing the 2.6.18 source from ftp.kernel.org and unpacking it finds the following contents: #ifndef _LINUX_CONFIG_H #define _LINUX_CONFIG_H /* This file is no longer in use and kept only for backward compatibility. * autoconf.h is now included via -imacros on the commandline */ #include linux/autoconf.h #endif but it doesn't exist in the FC6 kernel tree for 2.6.18-1.2798.fc6 ... Likewise, attempting to build Asterisk-1.2.13 fails thusly: chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/root/asterisk/asterisk-1.2.13/channels' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk-1.2.13]# and the path: /usr/include/linux doesn't have a compiler.h :o( Hacking at the Makefile I added: INCLUDE+=-I/usr/src/kernels/2.6.18-1.2798.fc6-i686 which gets us a compiler.h and the build process continues somewhat further but then blows up due to a lack of zaptel.h (back to first problem) My box is acient, having been RedHat 9, then FC2, FC3, FC5 and now FC6 ... so just to prove I'm not going mad (or possibly more likely the progressive upgrades left something broken) I did a clean install of FC6 to a spare box and tried to build with the same results. So, to a bit more hacking... I copied "config.h" from the clean 2.6.18 kernel source tree to the FC6 tree in inlcude/linux and now I can compile and install the zaptel stuff, but I can't load it: [EMAIL PROTECTED] zaptel-1.2.10]# modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.18-1.2798.fc6/misc/zaptel.ko): Invalid module format I've copied the compiler.h across to /usr/inlcude/linux and now I can compile and link Asterisk-1.2.13 along with Asterisk-Addons-1.2.5 so I am able to get my system back online but only by removing the zaptel configuration :o( Is Asterisk using deprecated kernel header files that some distros have decided to stop supporting/shipping? What do I need to do to get Zaptel drivers correctly compiled for my FC6 box? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk does not bridge zap channels on outgoing calls
Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring Dial Zap/g5/9399||T 00:07:58 (None) Obviously, this is a big problem for us... Below are my zapata.conf, zaptel.conf and extensions.conf: -- zapata.conf -- [channels] usecallerid=yes hidecallerid=no echocancel=yes musiconhold=service busydetect=yes ;callprogress=yes busycount=3 flash=20 rxflash=40 transfer=yes threewaycalling=yes ;rxgain=100% ;txgain=1.0 ;relaxdtmf=yes ;-- context=int_soft ;-- group=1 callgroup=1 pickupgroup=1,6,7 threewaycalling=yes transfer=yes ;useincomingcalleridonzaptransfer=yes callwaitingcallerid=yes ;echocancelwhenbriged=yes immediate=no rxgain=-2.0 txgain=2.2 signalling=fxo_ks callerid=Soft 1 channel=1 callerid=Soft 2 channel=2 callerid=Soft 3 channel=3 callerid=Soft 4 channel=4 callerid=Soft 5 channel=5 callerid=Soft 6 channel=6 callerid=Soft 7 channel=7 ;-- context=int_omg ;-- group=2 callgroup=2 pickupgroup=2,5,6 threewaycalling=yes transfer=yes ;useincomingcalleridonzaptransfer=yes callwaitingcallerid=yes ;echocancelwhenbriged=yes immediate=no rxgain=-2.0 txgain=2.2 callerid=OMG 28 channel=8 callerid=OMG 29 channel=9 callerid=OMG 30 channel=10 callerid=OMG 31 channel=11 callerid=OMG 32 channel=12 callerid=OMG 33 channel=13 callerid=OMG 34 channel=14 callerid=OMG 35 channel=15 callerid=OMG 36 channel=16 ;--- ;Placa TDM24XXP - 24 DE INTERIOARE; CONTEXT NOU!! ;--- ;-- context=int_agentie ;-- group=3 callgroup=3 pickupgroup=3,4,5,10 threewaycalling=yes transfer=yes ;useincomingcalleridonzaptransfer=yes callwaitingcallerid=yes ;echocancelwhenbriged=yes immediate=no rxgain=-2.0 txgain=2.2 callerid=Agentie 45 channel=25 callerid=Agentie 46 channel=26 callerid=Agentie 47 channel=27 callerid=Agentie 48 channel=28 callerid=Agentie 49 channel=29 callerid=Agentie 50 channel=30 callerid=Agentie 51 channel=31 callerid=Agentie 52 channel=32 callerid=Agentie 53 channel=33 callerid=Agentie 54 channel=34 callerid=Agentie 55 channel=35 callerid=Agentie 56 channel=36 callerid=Agentie 57 channel=37 callerid=Agentie 58 channel=38 callerid=Agentie 59 channel=39 callerid=Agentie 60 channel=40 callerid=Agentie 61 channel=41 callerid=Agentie 62 channel=42 callerid=Agentie 63 channel=43 callerid=Agentie 64 channel=44 callerid=Clopotel 65 channel=45 callerid=Clopotel 66 channel=46 callerid=Clopotel 67 channel=47 callerid=Clopotel 68 channel=48 ;-- context=cap_hunting ;-- group=4 callgroup=4 callerid=asreceived cidsignalling=v23 cidstart=ring transfer=yes threewaycalling=yes immediate=no useincomingcalleridonzaptransfer=yes ;echocancelwhenbriged=yes ;musiconhold=guitar sendcalleridafter=2 rxgain=10.2 txgain=1.8 signalling=fxs_ks channel=17 ;-- context=omegasoft ;-- group=10 callgroup=10 callerid=asreceived cidsignalling=v23 cidstart=ring transfer=yes ;threewaycalling=yes immediate=no useincomingcalleridonzaptransfer=yes ;echocancelwhenbriged=yes ;musiconhold=guitar sendcalleridafter=2 rxgain=10.2 txgain=1.8 signalling=fxs_ks channel=18 ;-- context=agentie ;-- group=5 callgroup=5 callerid=asreceived cidsignalling=v23 cidstart=ring transfer=yes threewaycalling=yes ;immediate=no ;useincomingcalleridonzaptransfer=yes ;echocancelwhenbriged=yes ;musiconhold=guitar sendcalleridafter=2 rxgain=10.2 txgain=1.8 signalling=fxs_ks channel=19-24 ;-- context=tehnic ;-- group=6 callgroup=6 callerid=asreceived cidsignalling=v23 cidstart=ring transfer=yes threewaycalling=yes ;immediate=no useincomingcalleridonzaptransfer=yes ;callprogres=yes ;echocancelwhenbriged=yes rxgain=10.2 txgain=1.8 signalling=fxs_ks channel=80 ;-- context=service ;-- group=7 callgroup=7 ;echocancelwhenbriged=yes callerid=asreceived answeronpolarityswitch=yes hanguponpolarityswitch=yes ;cidstart=ring ;cidsignalling=v23 sendcalleridafter=2 rxgain=5.5 txgain=7.2 signalling=fxs_ks channel=81 ;-- context=service_out ;-- group=7 callgroup=7 ;echocancelwhenbriged=yes callerid=asreceived ;cidstart=ring ;cidsignalling=v23 answeronpolarityswitch=yes hanguponpolarityswitch=yes sendcalleridafter=2 rxgain=4.5 txgain=5.2
[asterisk-users] Bridging Video Calls using Zap
Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as switch. On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile operator. What I have tried so far ist to use SetTransferCapability(VIDEO) but this does not change the behavior. Is there a way to set or preserve the bearer capability for the outgoing call leg? cheerio Steve --- environment pbx-test*CLI show version Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q built by root @ pbx-test.bb.ic3s.de on a i686 running Linux on 2006-06-16 10:17:00 UTC with A Quad ZAP Pri Card dialplan ;exten = 297,1,SetTransferCapability(VIDEO) exten = 297,1,Noop() exten = 297,2,Dial(${TRUNK}/0175234567) exten = 297,3,Hangup exten = 297,104,SetVar(PRI_CAUSE=17) ; Indicate Busy exten = 297,105,Hangup Log pbx-test*CLI == Primary D-Channel on span 2 down Oct 31 09:05:03 WARNING[6771]: chan_zap.c:2506 pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway! 1 Protocol Discriminator: Q.931 (8) len=45 1 Call Ref: len= 2 (reference 25880/0x6518) (Originator) 1 Message type: SETUP (5) 1 [1 a11 ] 1 Sending Complete (len= 1) 1 [1 041 031 881 901 a61 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: G.7xx 384k Video (38) 1 [1 181 031 a11 831 871 ] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 1 ChanSel: Reserved 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 7 ] 1 [1 6c1 0c1 211 831 311 371 351 351 381 361 341 371 381 311 ] 1 Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation allowed of network provided number (3) '1755864781' ] 1 [1 701 081 c11 351 351 351 361 321 391 371 ] 1 Called Number (len=10) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5556297' ] 1 [1 7c1 031 881 901 a61 ] 1 Low-layer compatibilty (len= 5) [ 1 0x88 1 0x90 1 0xA6 1 ] 1 -- Making new call for cr 25880 1 -- Processing Q.931 Call Setup 1 -- Processing IE 161 (cs0, Sending Complete) 1 -- Processing IE 4 (cs0, Bearer Capability) 1 -- Processing IE 24 (cs0, Channel Identification) 1 -- Processing IE 108 (cs0, Calling Party Number) 1 -- Processing IE 112 (cs0, Called Party Number) 1 -- Processing IE 124 (cs0, Low-layer Compatibility) Oct 31 09:05:07 WARNING[6770]: chan_zap.c:8503 zt_pri_error: 1 copying 5 bytes LLC 1 Protocol Discriminator: Q.931 (8) len=10 1 Call Ref: len= 2 (reference 25880/0x6518) (Terminator) 1 Message type: CALL PROCEEDING (2) 1 [1 181 031 a91 831 871 ] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 1 ChanSel: Reserved 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 7 ] -- Accepting data call from '1755864781' to '5556297' on channel 0/7, span 1 -- Executing Macro(Zap/7-1, handle-callerid) in new stack -- Executing NoOp(Zap/7-1, 01751234567) in new stack -- Executing GotoIf(Zap/7-1, 0?3:5) in new stack -- Goto (macro-handle-callerid,s,5) -- Executing SetCallerID(Zap/7-1, 001751234567) in new stack -- Executing LookupCIDName(Zap/7-1, ) in new stack -- Changed Caller*ID name to Testi Tester -- Executing NoOp(Zap/7-1, Testi Tester 001751234567 / Testi Tester / 001751234567) in new stack -- Executing Goto(Zap/7-1, external-call|297|1) in new stack -- Goto (external-call,297,1) -- Executing Dial(Zap/7-1, Zap/r1/01752345678) in new stack 1 -- Making new call for cr 32777 -- Requested transfer capability: 0x08 - DIGITAL 1 Protocol Discriminator: Q.931 (8) len=44 1 Call Ref: len= 2 (reference 9/0x9) (Originator) 1 Message type: SETUP (5) 1 [1 041 021 881 901 ] 1 Bearer Capability (len= 4) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 0 User information layer 1: Unknown (24) 1 [1 181 031 a91 831 881 ] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 1 ChanSel: Reserved 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 8 ] 1 [1 6c1
[asterisk-users] Read cmd
Hi list, I have been using the read command and I have noticed that it behaves more like the playback command and not background. Is there any way to set it up that I can enter a selection before Asterisk finishes playing the file ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting up UTStarcom F300
Hi List, I have a F300. I have not yet been able to set it up. Can anyone email me the exact configs on how they set it up to work with Asterisk ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] light web user interface
What attributes are you talking about ? Depending on what they are it may be real simple to set something up. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Monday, October 30, 2006 9:51 PM Subject: [asterisk-users] light web user interface Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Architecture for Asterisk
31 okt 2006 kl. 01.37 skrev je .: Thanks for the diagram. Is it possible to get a more detailed diagram. I'm looking for something a little more technical. In other words, where does Asterisk stand when inviting a user, when hanging up, when canceling an invitation etc.. Does it go direct from user to user or does it go to Asterisk? Asterisk is not a SIP proxy, we're a back-to-back-SIP-ua, b2bua. Everything ends in Asterisk and Asterisk, being a PBX, decides what to do next. In some cases we re-invite the media to go p2p, but SIP signalling ends in Asterisk. In SIP terminology: We're a SIP registrar/location server, SIP outbound proxy (or can be configured as one to be more correct) and a SIP b2bua, but not a SIP proxy. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Architecture for Asterisk
31 okt 2006 kl. 06.34 skrev je .: Thank you for the link. Chapter 8 was most useful in explaining the different types of connections (user/peer/friend) as well as the register function such that users may know how to contact another user. However I'm looking for something more specific. For instance, for a normal session termination (i.e. BYE), user1 would send msg BYE to proxy who would then forward it to user2 who would then close the connection. For a cancel request the following messages are exchanged: u1 - proxy: invite proxy - u2: arp request In this scenario for instance (where user 2 closes the connection), where does Asterisk fit in? If at all? Does Asterisk behave as the proxy? In an Asterisk scenario, you would have two different calls, one from u1 to Asterisk, one from Asterisk to u2. We never forward any SIP messages, like a SIP proxy. If u1 hangs up, we decide to hang up our call to u2, but those are two different SIP dialogs. Remember that Asterisk is a multiprotocol PBX. The connection to U2 might be using a different signalling protocol, like ISDN PRI. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Next training: Stockholm, Sweden, November 13th! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple dial macros at the same time
When user A is bridged with the client (right before it is done) set a variable that the call was taken, this way when he presses 1 to accpet the system checks to see what the value is, if its in taken mode he gets the message and then gets hung up on. - Original Message - From: Graham Mainwaring [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 3:38 AM Subject: [asterisk-users] Multiple dial macros at the same time I am setting up an after-hours on-call system. Someone calls in and requests service, and while they listen to music on hold, we dial out to several people's cell phones and home phones. We don't know if they will be answered by the employee, or by voicemail or a spouse/relative/child/pet. So we play a message that says press 1 to accept the call and ask employees to train their spouse/relative/child/pets not to press 1. The following extract from my dialplan shows how I have this feature set up. This is with Asterisk 1.2.13. [macro-screen] exten = s,1,Set(MACRO_RESULT=CONTINUE) exten = s,2,Read(ACCEPT|press-1-to-accept|1|skip|3|1) exten = s,3,GotoIf($[${ACCEPT}=1]?5:4) exten = s,4,MacroExit exten = s,5,Set(MACRO_RESULT=) exten = s,6,Playback(please-say-hello) [menu] exten = _FOLS1NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20,oM(screen)) exten = s,1,Playback(welcome) exten = s,2,Dial(LOCAL/FOLS19195551000LOCAL/FOLS19195552000,,tm) exten = s,3,Voicemail(u301) exten = s,4,Hangup In order for this to work, I needed the ability to restore MACRO_RESULT back to an unset state. For now I just hacked the Set application so that after removing the variable from the context, it only re-creates it if the value provided is greater than zero length. In the future I will probably write an UnSet application to handle this more gracefully, unless someone knows a better way to unset a variable. This all works fine, with one small problem that is driving me batty. I would appreciate any insight or ideas on how to solve this. Here's the scenario: 1. Caller dials the number and hears the welcome message, then music on hold. 2. Simultaneous calls are made to Employee A at 555-1000 and Employee B at 555-2000 (per above). 3. Both of them answer the phone. 4. Employee A presses one and hears you will now be connected, please say hello to the caller. 5. Employee A is bridged to the caller, says hello, and begins working with them. 6. A few seconds later, the Employee B also presses 1. He also hears you will now be connected. 7. Employee B fails to bridge, and is hung up on. The problem is, these are pretty urgent calls and employees are highly motivated to make sure they get answered. Employee B doesn't know whether the call dropped because someone else got it, or because of a phone system problem of some sort. He is now obligated to figure out what's up with the call and make sure someone got it. What I want instead is for Employee B to hear an alternate message that says someone else got the call. This gives positive confirmation that it's not his problem, so he can roll over and go back to sleep. I can see two ways of doing this. 1. Write a function called BridgedChannel that takes a channel ID and returns its bridge peer channel ID, if any. This would allow me to set a variable __PARENTCHANNEL with the channel ID of the incoming call, before the Dial command executes. The macro, at priority 6, can then check BridgedChannel(${__PARENTCHANNEL}). If it has a value then the call is already bridged and we can tell the employee not to worry. 2. Have a MySQL database with a single table with two fields, varchar channel-ID and boolean answered. When the call starts do update table set answered=false where channel-ID=${__PARENTCHANNEL}. When an employee dials 1, retrieve the value of answered for __PARENTCHANNEL and also set it to true in a single transaction. If the returned value was false, tell them to answer the call and bridge; if the returned value was true, tell them to go back to sleep and hang up. Solution #1 requires me to write a whole new function, and solution #2 requires a MySQL database, which is pretty big dependency for such a simple function. Does anyone see a simpler way of doing this, or have any ideas for other avenues to pursue? Thanks in advance, -Graham ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] best gui
Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: IAX2 show peers - description
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I think the (T) is for Trunk. Regards Fred Hi Fred! I believe that T is for trunk. Thank you. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time
Brad Templeton wrote: I've read a lot of the descriptions of handling NAT with Asterisk, and the use of both the nat and canreinvite flags. I am very familiar with Sip and NAT but have not seen an answer to the following question. My Asterisk server runs on a machine with two ethernets. One is an external net, with exposed IP addresses. The other is an internal net with natted IP addresses. Thus the server has two addresses. The server is _not_ the NAT gateway. That's a linksys box which has its own external IP to gateway traffic from the internal natwork. The phones are on the internal NATwork. Asterisk talks to them over it. Outside peers, such as SIP termination providers etc. talk to the Asterisk server via its outside address, which is as you would expect. However, from time to time I get the famous one-way audio because Asterisk has decided to do a native bridge between a natted SIP phone and an external SIP peer. It sends the internal IP of the SIP phone in the SDP and of course the outside service can't send packets to that. I could just turn off reinvites on the internal phones, but this would cause them to route all traffic through the asterisk box, even on internal calls between phones on the same ethernet, which seems foolish to me. I don't want to turn off reinvites to the external peers -- if a call comes in from a SIP originator for example, and is send back out to a SIP terminator (call forwarding) I want a native bridge for sure.(Handling the internal traffic is not so much of a burden though sometimes I hear latency because of it, but routing external traffic through the asterisk box is a bad thing.) So what I want is for Asterisk to use native bridges when connecting two channels behind the NAT, or two channels on the real internet, but not to do so when connecting an internal and external channel. It should be able to see the IP addresses, and know the difference between natted and external ones and know they can't talk to one another. (The ICE protocol would handle this someday.) Have you tried setting the externalip and localnet parameters? Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue
Rajkumar S wrote: -- Executing Queue(SIP/1002-74e9, Auth-Enq|t) in new stack -- Started music on hold, class 'default', on channel 'SIP/1002-74e9' -- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1001||tS(30)) in new stack -- Setting call duration limit to 30 seconds. -- Called 1001 -- Called Agent/1001 -- SIP/1001-d43c is ringing -- Agent/1001 is ringing -- SIP/1001-d43c answered Local/[EMAIL PROTECTED],2 -- Agent/1001 answered SIP/1002-74e9 -- Stopped music on hold on SIP/1002-74e9 == Spawn extension (from-sip, 1001, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (from-sip, 99, 1) exited non-zero on 'SIP/1002-74e9' Someone correct me if I'm wrong: The Dial string is missing a '/n' parameter for the Local channel. Without /n, Asterisk will do a native transfer to SIP/1001 and lose the timeout value defined earlier. Leo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dropping extra frame of G.729 since we already have a VAD frame at the end
Hello, I have a problem with a new provider and perhaps you can help me . whenI send traffic to this new provider i have this error : Dropping extra frame of G.729 since we already have a VAD frame at the end and the quality of the voice is bad because some parts of words are dropped. Laurent ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva server 4bri and Portuguese BRI
On Tue, 31 Oct 2006, Pedro Silva wrote: Hello, One problem is solved and another appears... :( I cannot receive incoming calls on trixbox. I defined one incoming route (any DID/any CID) and forwading these calls to a SIP extension. With capi and sip debug in asterisk -r console i dont detect any incoming activity... Did you use set verbose 5 capi debug ? If not, you should see anything there. But if you don't see activity with this verbose level too, this call is not signaled through capi. In that case you should create traces with divactrl ditrace (or the trace wizard) to get capi activity too. Armin In xlog console i have the following debug: 0:1898:127 - SIG-R(034) 08 01 0D 05 A1 04 03 80 90 A3 18 01 81 6C 0B 00 83 39 36 33 30 34 35 37 32 33 70 02 81 30 7D 02 91 81 Q.931 CR0d SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 81 Calling Party Number 00 83 '963045723' Called Party Number 81 '0' HLC 91 81 0:1898:127 - SIG-S 0-6 e:805 0:1898:130 - CREATEID ok: context:1f assigned Id:3 freeIds=ec 0:1898:130 - alloc cr in use =4 0:1898:133 - SIG-X(008) 08 01 8D 45 08 02 80 95 Q.931 CR8d DISC Cause 80 95 'Call rejected' 0:1898:133 - SIG-x(008) 08 01 8D 5A 08 02 80 D8 Q.931 CR8d REL_COM Cause 80 d8 'Incompatible destination' 0:1898:133 - SIG-S 6-0 e:8c5 0:1898:134 - D-X(012) 00 01 14 16 08 01 8D 5A 08 02 80 D8 0:1898:135 - free cr in use =3 0:1898:135 - DELETEID ok: deleted Id:4 freeIds=ec 0:1898:155 - D-R(004) 00 01 01 16 So the problem appears to be Incompatible destination... but is problem in asterisk or is before asterisk, on diva card...? Tanks by any possible help! Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Finally this works!!! :) Tanks to Alberto and Marco by your help! The problems are: - the cable was connected to the wong card port... :( - the card config needs to be: ETSI; TE; Point-to-Point (I thought that was point-to-multipoint). Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Hello again Alberto! Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.com) This is the xlog output: 4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:074 - alloc cr in use =4 4:1736:076 - free cr in use =3 4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:078 - alloc cr in use =4 4:1736:080 - free cr in use =3 4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:081 - alloc cr in use =4 4:1736:083 - [1,0] dsp_assign 0016, 0, 2048 4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08 4:1736:084 - [1,0] Download 532 requested 4:1736:084 - MORE 4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E 02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33 Q.931 CR36 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 83 Progress Indicator 80 83 Called Party Number 80 '963045723' 4:1736:085 - SIG-S 0-1 e:885 4:1736:087 - ACTIVATION_REQ 4:1744:147 - L1_DOWN 4:1744:150 - SIG-EVENT 08 4:1744:150 - SIG-EVENT 08 4:1744:150 - EVENT: Call failed in State 'Call initiated' Link disconnected, Layer-1 error (cable or NT) 4:1744:150 - SIG-S 1-0 e: 4:1744:151 - [1,0] dsp_release 4:1744:155 - free cr in use =3 4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb I disconnect the rj45 cable from alcatel pbx and connect that to the diva card (with alcatel pbx i can make calls normally). The green led of the diva card is activated when i connect the cable. So i dont understand why the message Link disconnected, Layer-1 error (cable or NT)... This debug is th same if the cable is connected to the NT or not. Any ideas...? Thanks! PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva server 4bri and Portuguese BRI
Hi Pedro, pls post your capi.conf! I'm not used with CAPI, but should have something like: [interfaces] incomingmsn=* ; Here you match MSNs arriving from telco, for debug let it ' * ' controller=1 softdtmf=1 accountcode= context=demo ;Set this to ext-did that should be the context TRIXBOX will handle By the way, Trybox probably has an extensions_custom.conf In this file try this: [ext-did-custom] exten=_X.,1,Answer exten= _X.,n,Noop(Debugging MSNs from Telco: ${EXTEN}) exten= _X.,n,wait(1) exten= _X.,n,playback(tt-monkeys) exten=_X.,n,hangup Hope this helps. Pls give some feedback On 10/31/06, Armin Schindler [EMAIL PROTECTED] wrote: On Tue, 31 Oct 2006, Pedro Silva wrote: Hello, One problem is solved and another appears... :( I cannot receive incoming calls on trixbox. I defined one incoming route (any DID/any CID) and forwading these calls to a SIP extension. With capi and sip debug in asterisk -r console i dont detect any incoming activity... Did you use set verbose 5 capi debug ? If not, you should see anything there. But if you don't see activity with this verbose level too, this call is not signaled through capi. In that case you should create traces with divactrl ditrace (or the trace wizard) to get capi activity too. Armin In xlog console i have the following debug: 0:1898:127 - SIG-R(034) 08 01 0D 05 A1 04 03 80 90 A3 18 01 81 6C 0B 00 83 39 36 33 30 34 35 37 32 33 70 02 81 30 7D 02 91 81 Q.931 CR0d SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 81 Calling Party Number 00 83 '963045723' Called Party Number 81 '0' HLC 91 81 0:1898:127 - SIG-S 0-6 e:805 0:1898:130 - CREATEID ok: context:1f assigned Id:3 freeIds=ec 0:1898:130 - alloc cr in use =4 0:1898:133 - SIG-X(008) 08 01 8D 45 08 02 80 95 Q.931 CR8d DISC Cause 80 95 'Call rejected' 0:1898:133 - SIG-x(008) 08 01 8D 5A 08 02 80 D8 Q.931 CR8d REL_COM Cause 80 d8 'Incompatible destination' 0:1898:133 - SIG-S 6-0 e:8c5 0:1898:134 - D-X(012) 00 01 14 16 08 01 8D 5A 08 02 80 D8 0:1898:135 - free cr in use =3 0:1898:135 - DELETEID ok: deleted Id:4 freeIds=ec 0:1898:155 - D-R(004) 00 01 01 16 So the problem appears to be Incompatible destination... but is problem in asterisk or is before asterisk, on diva card...? Tanks by any possible help! Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Finally this works!!! :) Tanks to Alberto and Marco by your help! The problems are: - the cable was connected to the wong card port... :( - the card config needs to be: ETSI; TE; Point-to-Point (I thought that was point-to-multipoint). Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Hello again Alberto! Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.com) This is the xlog output: 4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:074 - alloc cr in use =4 4:1736:076 - free cr in use =3 4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:078 - alloc cr in use =4 4:1736:080 - free cr in use =3 4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:081 - alloc cr in use =4 4:1736:083 - [1,0] dsp_assign 0016, 0, 2048 4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08 4:1736:084 - [1,0] Download 532 requested 4:1736:084 - MORE 4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E 02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33 Q.931 CR36 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 83 Progress Indicator 80 83 Called Party Number 80 '963045723' 4:1736:085 - SIG-S 0-1 e:885 4:1736:087 - ACTIVATION_REQ 4:1744:147 - L1_DOWN 4:1744:150 - SIG-EVENT 08 4:1744:150 - SIG-EVENT 08 4:1744:150 - EVENT: Call failed in State 'Call initiated' Link disconnected, Layer-1 error (cable or NT) 4:1744:150 - SIG-S 1-0 e: 4:1744:151 - [1,0] dsp_release 4:1744:155 - free cr in use =3 4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb I disconnect the rj45 cable from alcatel pbx and connect that to the diva card (with alcatel pbx i can make calls normally). The green led of the diva card is activated when i connect the cable. So i dont understand why the message Link disconnected, Layer-1 error (cable or NT)... This debug is th same if the cable is connected to the NT or not. Any ideas...? Thanks! PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] sip realtime broken?
Can someone confirm if sip realtime is broken in 1.2.13 and if so when was the last release it wasn't? heh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip realtime broken?
i've been using it sucessfully On 10/31/06, Don [EMAIL PROTECTED] wrote: Can someone confirm if sip realtime is broken in 1.2.13 and if so when was the last release it wasn't? heh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dial D option with w for wait?
From WIKI: D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) When I use the D option to send a call to my paging system and pick a zone, the Tone is too early. I have tried the 'w' option, but it does not appear to work. No matter how many 'w's I use, the tone is still immediately on answer. Is this a known issue? Is there a work around? My current workaround is to send both channel to a meetme that runs a macro to play the tone. This is way to much overhead to play a single tone after .5 or 1 seconds. Please advise. Thank You, Steven BerkHolz Board member of www.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridging Video Calls using Zap
Hi, You have to modify the setup sent for the call outgoing... Regards, Tristan Steffen Weinreich a écrit : Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as switch. On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile operator. What I have tried so far ist to use SetTransferCapability(VIDEO) but this does not change the behavior. Is there a way to set or preserve the bearer capability for the outgoing call leg? cheerio Steve --- environment pbx-test*CLI show version Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q built by root @ pbx-test.bb.ic3s.de on a i686 running Linux on 2006-06-16 10:17:00 UTC with A Quad ZAP Pri Card dialplan ;exten = 297,1,SetTransferCapability(VIDEO) exten = 297,1,Noop() exten = 297,2,Dial(${TRUNK}/0175234567) exten = 297,3,Hangup exten = 297,104,SetVar(PRI_CAUSE=17) ; Indicate Busy exten = 297,105,Hangup Log pbx-test*CLI == Primary D-Channel on span 2 down Oct 31 09:05:03 WARNING[6771]: chan_zap.c:2506 pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway! 1 Protocol Discriminator: Q.931 (8) len=45 1 Call Ref: len= 2 (reference 25880/0x6518) (Originator) 1 Message type: SETUP (5) 1 [1 a11 ] 1 Sending Complete (len= 1) 1 [1 041 031 881 901 a61 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: G.7xx 384k Video (38) 1 [1 181 031 a11 831 871 ] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 1 ChanSel: Reserved 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 7 ] 1 [1 6c1 0c1 211 831 311 371 351 351 381 361 341 371 381 311 ] 1 Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation allowed of network provided number (3) '1755864781' ] 1 [1 701 081 c11 351 351 351 361 321 391 371 ] 1 Called Number (len=10) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5556297' ] 1 [1 7c1 031 881 901 a61 ] 1 Low-layer compatibilty (len= 5) [ 1 0x88 1 0x90 1 0xA6 1 ] 1 -- Making new call for cr 25880 1 -- Processing Q.931 Call Setup 1 -- Processing IE 161 (cs0, Sending Complete) 1 -- Processing IE 4 (cs0, Bearer Capability) 1 -- Processing IE 24 (cs0, Channel Identification) 1 -- Processing IE 108 (cs0, Calling Party Number) 1 -- Processing IE 112 (cs0, Called Party Number) 1 -- Processing IE 124 (cs0, Low-layer Compatibility) Oct 31 09:05:07 WARNING[6770]: chan_zap.c:8503 zt_pri_error: 1 copying 5 bytes LLC 1 Protocol Discriminator: Q.931 (8) len=10 1 Call Ref: len= 2 (reference 25880/0x6518) (Terminator) 1 Message type: CALL PROCEEDING (2) 1 [1 181 031 a91 831 871 ] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 1 ChanSel: Reserved 1Ext: 1 Coding: 0 Number Specified Channel Type: 3 1Ext: 1 Channel: 7 ] -- Accepting data call from '1755864781' to '5556297' on channel 0/7, span 1 -- Executing Macro(Zap/7-1, handle-callerid) in new stack -- Executing NoOp(Zap/7-1, 01751234567) in new stack -- Executing GotoIf(Zap/7-1, 0?3:5) in new stack -- Goto (macro-handle-callerid,s,5) -- Executing SetCallerID(Zap/7-1, 001751234567) in new stack -- Executing LookupCIDName(Zap/7-1, ) in new stack -- Changed Caller*ID name to Testi Tester -- Executing NoOp(Zap/7-1, Testi Tester 001751234567 / Testi Tester / 001751234567) in new stack -- Executing Goto(Zap/7-1, external-call|297|1) in new stack -- Goto (external-call,297,1) -- Executing Dial(Zap/7-1, Zap/r1/01752345678) in new stack 1 -- Making new call for cr 32777 -- Requested transfer capability: 0x08 - DIGITAL 1 Protocol Discriminator: Q.931 (8) len=44 1 Call Ref: len= 2 (reference 9/0x9) (Originator) 1 Message type: SETUP (5) 1 [1 041 021 881 901 ] 1 Bearer Capability (len= 4) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 0 User information layer 1: Unknown (24) 1 [1 181 031 a91 831 881 ] 1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 1 ChanSel: Reserved 1
[asterisk-users] Re: Forwarding recorded calls to Voicemail
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I was wondering if anyone has implemented a feature that would allow a user to record a phone call and once the call has ended, the call is forwarded to his voicemail? Hi Tom! I was looking for something like this, but I was unable to find anything useful. Hopefully someone will answer your mail. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip realtime broken?
not the sip.conf static realtime but sipusers etc? - Original Message - From: Marco Mouta [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 31, 2006 7:26 AM Subject: Re: [asterisk-users] sip realtime broken? i've been using it sucessfully On 10/31/06, Don [EMAIL PROTECTED] wrote: Can someone confirm if sip realtime is broken in 1.2.13 and if so when was the last release it wasn't? heh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.17/505 - Release Date: 10/27/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] light web user interface
Basically I would like a page that would allow a user to log in and modify their extension only. So for example, I log in for extension 102 once in there I can turn on or off my call waiting. Add a number to call forward to. Change the email address my voice mail gets sent to. Add any numbers I may want to block via caller ID. Maybe view my voice mails that are saved and be able to download them in wav format from there. Add find me follow me extensions and numbers, etc I would also like it open enough that I can add features to it. Im not the best at PHP but I can work my way around in it. I thought maybe freePBX allowed this with its users but I cant see where you can lock them down to only see information on a particular extension. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, October 31, 2006 3:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] light web user interface What attributes are you talking about ? Depending on what they are it may be real simple to set something up. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Monday, October 30, 2006 9:51 PM Subject: [asterisk-users] light web user interface Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] +Ura +md3200 nao encaminha ligacao
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Did you ignore that is an english discussion list or did you just post on the wrong list altogether? Anyhoo, try relaxdtmf=yes in your zapata.conf. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFR1P+2QVs8jsa1mQRAnE9AKCNj8pK4EEFx8TWQFuLXXIH+TbRawCfXB/S 7sbNds3FrP8tnNQyb++YbJw= =w56E -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP RTP flow
Hey, This is probably a rather stilly question... If I pick up my SIP phone that's registered to my asterisk server and dial a number that asterisk recognises as destined for a SIP trunk (could be a static route, or an ENUM lookup) or another SIP device registered on said asterisk server (internal extension to extension call), what route does the actual audio take? The control connection (port 5060) obviously goes via the asterisk server as it has to work out where to send the control to, but I could quite easily imagine the audio going directly handset to remote server or handset to asterisk to remote, and handset to handset or handset to asterisk to handset. Thanks -- Mike Williams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk dial out (in SIP) to another asterisk context !
Hello list, We have 2 asterisk servers (without firewall and NAT), and We want to do : From the first server, we have a .call file which dial out to the second server. The second server automatically answers and Play a music during X seconds, then it hangs up. Is it possible? Thanks you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem
Anybody knows why ARI gives this error message when I enter extension number and password. Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed to open stream: Permission denied in /var/www/html/recordings/modules/voicemail.module on line 525 It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone
Which asterisk release are you running chan_skinny under? - Original Message - From: Will Roy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phone Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7] What additional information would be required to troubleshoot this? or should I stop wasting time and just convert the phone to SIP? :) regards Wil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP RTP flow
You can make RTP pass through Asterisk, or not. Look in voip-info.org about Native Bridge and sip.conf canreinvite option. And may be this page will be usefull too: http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy Regards On 10/31/06, Mike Williams [EMAIL PROTECTED] wrote: Hey, This is probably a rather stilly question... If I pick up my SIP phone that's registered to my asterisk server and dial a number that asterisk recognises as destined for a SIP trunk (could be a static route, or an ENUM lookup) or another SIP device registered on said asterisk server (internal extension to extension call), what route does the actual audio take? The control connection (port 5060) obviously goes via the asterisk server as it has to work out where to send the control to, but I could quite easily imagine the audio going directly handset to remote server or handset to asterisk to remote, and handset to handset or handset to asterisk to handset. Thanks -- Mike Williams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridging Video Calls using Zap
Tristan schrieb: Hi, You have to modify the setup sent for the call outgoing... OK, is there a way to this from the dialplan of have i to modify source for this? cheeiro Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem
User that web server is running has to have read permissions to file /var/spool/asterisk/voicemail/default/222/INBOX/msg.txt Easier option is to run apache as asterisk user Zeeshan Zakaria wrote: Anybody knows why ARI gives this error message when I enter extension number and password. *Warning*: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed to open stream: Permission denied in */var/www/html/recordings/modules/voicemail.module* on line *525* It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem
On Tue, 2006-10-31 at 09:55 -0500, Zeeshan Zakaria wrote: Anybody knows why ARI gives this error message when I enter extension number and password. Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed to open stream: Permission denied in /var/www/html/recordings/modules/voicemail.module on line 525 Are you sure about the file permissions? The file /var/spool/asterisk/voicemail/default/222/INBOX/msg txt must be permissions for the apache user or group. Try changing the ownership of the file. Using Debian will be like (apache group is called www-data): chown asterisk:www-data /var/spool/asterisk/voicemail/default/222/INBOX/msg Regards, It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk web interface is not parsing the PHP pages
Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . Thanks and Regards Alok Mohapatra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem
it told you:Permission DeniedCheck the permissions on that directory.On 10/31/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:Anybody knows why ARI gives this error message when I enter extension number and password. Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed to open stream: Permission denied in /var/www/html/recordings/modules/voicemail.module on line 525 It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A Zakaria ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on virtual machine
We have a centralized infrastructure where we deploy Asterisk servers in remote call centers for authentication and transcoding. SIP g729a calls are then sent over an MPLS VPN to a central Asterisk farm, from which calls aresent/received via PRI. To avoid placing two servers in each call center, one for Asterisk and another for Windows AD services, we have been playing with VMWare. Can anyone provide their experiences in using Asterisk in a VMWare configuration? Good/bad/ugly? Thanks, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] auto recording extensions
I would like to know how to record all calls on a queue. Anu good sugestions? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto recording extensions
If you did make samples you should see an example in queues.conf. By default it's commented out. - Original Message - From: Ed Nuñez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 31, 2006 5:31 PM Subject: [asterisk-users] auto recording extensions I would like to know how to record all calls on a queue. Anu good sugestions? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Call Statistics
Hi Folks, I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format, are there some open source aplication for Asterisk that be easier for use. Pls anything suggestion will be very appreciate. Thanks Rgds. -- Omar E.P.T - Certified Networking Professionals make better Connections! http://omarept.blogspot.com/ Usysnet Corp Open Source Solutions www.usysnet.com.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages
Did you install PHP?On 10/31/06, Alok Mohapatra [EMAIL PROTECTED] wrote: Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . Thanks and Regards Alok Mohapatra ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audiocodes MP-114 noise
Dear Jason,Please define better noisy? You talking echo issues? Is it on just your side or on the called party's side as well?This start happening immediately, or was the box working before and the problem just started?Also, a quick heads up, make sure before even beginning to troubleshoot an issue like this you do a factory reset to the unit and get the latest available firmware on it. Usually that fixes annoying issues like this.Thanks, Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 30, 2006, at 10:36 PM, Jason Kim wrote:It's noisy while talking.Any idea?Thanks in advance.JasonCheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates (http://voice.yahoo.com)___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages
If it is not parsing the index.php mean that you see the code in your browser, install php Alok Mohapatra wrote: Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . Thanks and Regards Alok Mohapatra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astricon followup
For the benefit of those outside of the USA or those unable to make it to Astricon; I wanted to send out this email. For those of you who attended Astricon in Dallas last week what was the one thing that you saw that made the trip worthwhile? (if we post enough information or comments it will be of benefit for those that didnt attend) For me personally it was the volume of neat add-on applications that the Asterisk community are developing; Over time Im hoping that this leads to something like AppExchange from Salesforce.com were people can choose from over 300+ applications or addons for SF. I really want to see more speech recognition applications but I think its great what Lumen-vox are doing. Id also like to see someone post some more modified ftp to text to speech http://www.voip-info.org/wiki/view/asterisk+at+home+festival+weather+configuration It doesnt need to be weather, how about Oil futures or wheat prices or score for the weekends games. Any text file accessible by FTP can be implemented into this script. Id like to see more. Im hoping that over time we can see even more to the point that people buy Asterisk just for the applications and we can quote the same price if not more than cisco because of these addon applications. Cheers, Dean www.Mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages
After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . The problem is probably that you didn't install PHP yum install php ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Call Statistics
omar parihuana wrote: Hi Folks, I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format, are there some open source aplication for Asterisk that be easier for use. Pls anything suggestion will be very appreciate. Thanks Rgds. Hi, If you have asterisk-addons, you can get all CDR's, which include all the above statistics, written to a MySQL or PGSQL database. It would then be very easy to get this on to a web page. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] channel.c: Unable to request channel ZAP
Hi All, I have one rather annoying problem...my PBX can work great for weeks, when suddenly I start receiving these messages when I try to make a zaptel call: Oct 31 13:52:47 NOTICE[15636] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) Oct 31 13:52:49 NOTICE[15648] channel.c: Unable to request channel ZAP/g1/247 I'm using Sangoma A104 card (with four E1 spans), and these problems are only occurring on the first two spans (which are connected to a legacy PBX) the second two spans, which are connected to the Telco, work perfectly. Even more: when these messages start to occur, I can hardly initiate any call via problematic two spans (1st and 2nd), where I can with no problem initiate a new call thru the unproblematic two spans (3rd and 4th). Restart of the Asterisk is the only cure so far Does anyone know what could possibly be the cause, or how could I troubleshot this problem? Regards. Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
re: [asterisk-users] Live creation of trunk groups
Well it works. If I have group=0 that includes all my channels, I can create group=1 which is a subset and a simple reload makes this g1 available to dial on that subset. Message: 12 Date: Mon, 30 Oct 2006 15:25:06 -0700 From: Alyed Tzompa [EMAIL PROTECTED] Subject: re: [asterisk-users] Live creation of trunk groups To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 My advice is to first make some tests to see if a reload is enough for Asterisk to read any group definitions change in zapata.conf, otherwise no on-the-fly change will work Alyed Return-Path: [EMAIL PROTECTED] Mon Oct 30 13:23:36 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Mon, 30 Oct 2006 13:23:36 -0700 Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Andre Courchesne ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061030/8b841866/attachment-0001.htm -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR
On Mon, Oct 30, 2006 at 06:54:40PM -0500, Vitalie Apostu wrote: Greetings, If somebody knows how to concatenate several .gsm files in one or create a macro and use with background() please reply. As simple as: cat file1.gsm file2.gsm both.gsm -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages
Alok, Two things: 1) You said you installed AMP. AMP has ceased development a while ago, but is survived by the FreePBX project. If you actually installed AMP and not FreePBX, I would suggest you get FreePBX running first. A lot of effort went into improving FreePBX from AMP. 2) You typically won't find much help for the GUIs from this list because the GUIs have their own mailing lists and forums. Try posting your question to FreePBX.org. You're more likely to get a response there. Alex On 10/31/06, Alok Mohapatra [EMAIL PROTECTED] wrote: Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . Thanks and Regards Alok Mohapatra ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk web interface is not parsing the PHPpages
Possibly a silly question, but do you have php installed and configured in apache? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alok MohapatraSent: 31 October 2006 15:45To: asterisk-users@lists.digium.comSubject: [asterisk-users] Asterisk web interface is not parsing the PHPpages Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . Thanks and Regards Alok Mohapatra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on virtual machine
What about the AD in the VM. or running Open LDAP on the Asterisk server. On 10/31/06, Adam Robins [EMAIL PROTECTED] wrote: We have a centralized infrastructure where we deploy Asterisk servers in remote call centers for authentication and transcoding. SIP g729a calls are then sent over an MPLS VPN to a central Asterisk farm, from which calls are sent/received via PRI. To avoid placing two servers in each call center, one for Asterisk and another for Windows AD services, we have been playing with VMWare. Can anyone provide their experiences in using Asterisk in a VMWare configuration? Good/bad/ugly? Thanks, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages
Hi Alok, it seems like libapache2-mod-php is missing in your linux box. Have you tried to make a simple index.php file to test? Giorgio Incantalupo Alok Mohapatra wrote: Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . Thanks and Regards Alok Mohapatra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem
Everytime a voicemail is recorded, a .txt file is generated. It was working fine before and permissions were automatically set. On my home server it is working perfectly fine. This is another server, with the same settings, and all of a sudden today it has started to give this error. Voicemails etc recorded yesterday are all fine, no problem with permissions. I don't remember changing anything on the server today which could have started giving this error. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom or Cisco Phones?
Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Example Polycom function key config
Hi, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example as I have tried various entries for hours now and don't seem to be getting anywhere. Any help appreciated. Kind regards Jamie Heckford Technical Consultant ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] simultaneous ring - call groups or queues or something else?
Hi, folks: I need to be able to have a single DID ring multiple remote (IP and PSTN) extensions, and then pass the call to whichever picks up first. I'm sure this is old hat -- lots of providers offer it. I see that Trixbox will do it, but it's not clear how it's doing it. They use different terminology -- a ring group and hunt strategy How can it be done with a straight Asterisk server? Thanks for the help! -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk on virtual machine
Asterisk does not work very well in a VM due to the timeslicing. Dropped calls, jittery audio and echo can all creep in. Good news is that an AD controller runs just fine in VMware. Just make sure the box has enough RAM to keep it happy, and use a physical second disk for the Windows install. So Id suggest running Asterisk in Linux as the native OS, and running VMware with Windows Server as a guest OS. This setup should work just fine for you. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Tuesday, October 31, 2006 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk on virtual machine We have a centralized infrastructure where we deploy Asterisk servers in remote call centers for authentication and transcoding. SIP g729a calls are then sent over an MPLS VPN to a central Asterisk farm, from which calls aresent/received via PRI. To avoid placing two servers in each call center, one for Asterisk and another for Windows AD services, we have been playing with VMWare. Can anyone provide their experiences in using Asterisk in a VMWare configuration? Good/bad/ugly? Thanks, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon followup
Hello All, This is a great list post, I have blogged about it here: http://www.asteriskvoipnews.com/asterisk_news/astricon_2006_followup.html It would be great if people could post there response on this post along with the list. I love reading answers to questions like this. Thanks, -Dal - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 31, 2006 8:44 AM Subject: [asterisk-users] Astricon followup For the benefit of those outside of the USA or those unable to make it to Astricon; I wanted to send out this email. For those of you who attended Astricon in Dallas last week what was the one thing that you saw that made the trip worthwhile? (if we post enough information or comments it will be of benefit for those that didnt attend) For me personally it was the volume of neat add-on applications that the Asterisk community are developing; Over time Im hoping that this leads to something like AppExchange from Salesforce.com were people can choose from over 300+ applications or addons for SF. I really want to see more speech recognition applications but I think its great what Lumen-vox are doing. Id also like to see someone post some more modified ftp to text to speech http://www.voip-info.org/wiki/view/asterisk+at+home+festival+weather+configuration It doesnt need to be weather, how about Oil futures or wheat prices or score for the weekends games. Any text file accessible by FTP can be implemented into this script. Id like to see more. Im hoping that over time we can see even more to the point that people buy Asterisk just for the applications and we can quote the same price if not more than cisco because of these addon applications. Cheers, Dean www.Mexuar.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] overlap of zap trunk groups
Can anyone suggest any reasons why a zap (PRI) b channel should not be a member of multiple zap trunk group definitions? For example; Group 1 = Channels 1 to 23 Group 2 = channels 1 to 12 Group 3 = channels 13 to 23 The purpose is to restrict the number of channels a particular extensions can use, but use the entire span for other extensions. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXO Cards vs. Channel bank with T1
Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time
On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote: Have you tried setting the externalip and localnet parameters? Localnet makes some sense, and is set (should be the default anyway, no?) externalip, as I understand it, is for an Asterisk which is behind a NAT. This asterisk is not behind a NAT to anybody. The phones are behind a NAT to the outside world but not to the Asterisk box, which has two ethernets on it, one for the internal natwork and one for the real internet. It uses bindaddr=0.0.0.0 and listens to both addresses. Sorry for my previous post I misunderstood the problem. You should set canreinvite=no to all sip peers that connect from outside. That's precisely what I don't want to do. This would block native bridging in the one case where it's most important. The correct behaviour, as I see it is: a) Native bridge when connecting two external channels -- everybody is on the real internet b) Native bridge when connecting two internal channels -- everybody is on the 192.168.* network c) Route RTP through Asterisk when connecting internal and external d) When a channel is to a device behind a remote NAT, the usual rules apply (either use STUN or other smart NAT, or route RTP through Asterisk) The super correct behaviour, which I don't expect but would be nice is e) Clever native bridge between internal and external by being aware that the device talks to the outside world using a different address than it talks to you. (Possibly if the phones use STUN they will tell Asterisk their external IP, which is not the same as Asterisk's though it's on the same subnet) I have used localnet=192.168.* and nat=yes on a local device and it still attempts an incorrect native bridge between internal and external, with one-way audio. If I do canreinvite=no on the local devices then it works of course, but now means the local phones will never native bridge amongst themselves. In a larger network, that would be a problem, and it's a poor result in any network. This is the latest svn of 1.2, by the way. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Snom or Cisco Phones?
Cisco Cisco or Linksys Cisco? Cisco Cisco, I'd prefer the Snom. Linksys Cisco, it's a tossup. I've worked with dozens of the Cisco 7960 phones, 25 of the Linksys, and 3 Snom. My specific issues with the Cisco included poor echo cancellation, problems with nat traversal, and no web interface. I didn't like any of the default ringers on the Snom phones, but the users really liked the LED call appearance lights compared to the 7960 LCD. I have no complaints about the Linksys phones. Ejay Hire -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Tuesday, October 31, 2006 11:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Snom or Cisco Phones? Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Recommendations
Hello, I'm working with supermicro servers, for the irq problems with Dell, any people have problems Regards On 10/30/06, Paul Hales [EMAIL PROTECTED] wrote: How many analog lines are you looking at? Hundreds?PaulHOn Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote: We have a number of clients who will be needing a server to host Asterisk on.Many of these clients use analog (FXO) lines that will need to be connected to Asterisk via Sangoma cards.Can anyone recommend an industry-standard server (like IBM, Dell, HP, etc.) that has enough open PCI slots to handle up to six of the Sangoma cards?We would like to be able to tell the customer to just go purchase this model server from this manufacturer and it will work.Suggestions? Thank you! Joe Dennick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simultaneous ring - call groups or queues or something else?
You can just seperate multiple phones with in the Dial command, as the voip-info wiki page shows: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial On Tue, Oct 31, 2006 at 10:28:32AM -0700, Stephen Bosch wrote: Hi, folks: I need to be able to have a single DID ring multiple remote (IP and PSTN) extensions, and then pass the call to whichever picks up first. I'm sure this is old hat -- lots of providers offer it. I see that Trixbox will do it, but it's not clear how it's doing it. They use different terminology -- a ring group and hunt strategy How can it be done with a straight Asterisk server? Thanks for the help! -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Characters in CLI on TTY9
When I look at TTY9 (using init.d and safe_asterisk to start the asterisk process), I am getting some strange characters. When a application is run the and the CLI shows the application executing the languange almost looks russian...?? Anyone seen this before? http://picasaweb.google.com/jonforrest.beck/AsteriskCLI ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones
I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct DTMF tones 25% of the time. I have to call several times to enter an extension. I have a router and a packet shaper and some other stuff. Anyone have any other ideas why this might happen. I do not have any Zap channels but I am running CentOS4. I also do not have any cards installed. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration problem
firewall? i dont think so because sometimes the phone can register ok and sudendly the appears unregistered Leonardo Silva [EMAIL PROTECTED] ha escrito: 2006/10/31, Jon Farmer [EMAIL PROTECTED]: Sergio R. D'Ippolito wrote: Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: */SIP/2.0 401 Unauthorized/* /Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x/ /From: SPA922 sip:[EMAIL PROTECTED];tag=685bbad1fae3325do0/ /To: SPA922 sip:[EMAIL PROTECTED];tag=as4da6f6ce/ /Call-ID: [EMAIL PROTECTED]/ /CSeq: 5503 REGISTER/ /User-Agent: incore-PBX/ /Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY/ /WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=372b2479/ Asterisk is asking the phone to resend the registration with WWW-Authenticate using MD5 hash. Make sure the phone supports this and retry. Or you could turn this option off in the sip.conf. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Maybe a Firewall ? -- Leonardo Silva fone: 16 8143-1146 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom or Cisco Phones?
Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia cars Joao Pereira wrote: Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards vs. Channel bank with T1
YES! Many machines do NOT work well with multiple analog cards. Especially the Digium ones. Channel banks with FXO circuits are harder to come by on the used market, though Many all FXS channel banks can be had used, though. If you want multiple FXO's and do not want to go the T1 route, look towards the Sangoma A200 John Novack Dovid B wrote: Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards vs. Channel bank with T1
Dovid B wrote: Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? Thanks. In my experience a T-1 port w/channel bank just works better. The more cards you use, the more interrupts are generated. My standard configuration for analog FXS ports is a T-1 card (Digium or Sangoma) and an Adtran TA750 Channel Bank. The Adtrans can be found very cheap on eBay. FXO ports tend to be much expensive, but you can find them on eBay as well. Why not just get a PRI or channelized voice T-1? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] overlap of zap trunk groups
Damon Estep wrote: Can anyone suggest any reasons why a zap (PRI) b channel should not be a member of multiple zap trunk group definitions? For example; Group 1 = Channels 1 to 23 Group 2 = channels 1 to 12 Group 3 = channels 13 to 23 The purpose is to restrict the number of channels a particular extensions can use, but use the entire span for other extensions. Part of a production /etc/asterisk/zaptel.conf: group=1 channel = 1-6 group=1,2 channel = 7-12 group=0 channel = 13-16 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards vs. Channel bank with T1
On Tue, Oct 31, 2006 at 08:20:57PM +0200, Dovid B wrote: Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? If you need enough ports to make a T-1 card cost-efficient, then you might oughtta be looking at an Ethernet to FXO media gateway instead -- assuming you need analog interfaces. FXO side, why not just go T-1 or PRI? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [asterisk-users] Snom or Cisco Phones?
I think one of the differences is: We do pay attention to Asterisk and this mailing list ;-) CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira Gesendet: Dienstag, 31. Oktober 2006 13:47 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Snom or Cisco Phones? Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel.c: Avoided initial deadlock
On 26 Oct 2006, at 13:25, asterisk wrote: Hi all, Can tell me somebody what meen : channel.c: Avoided initial deadlock Our customer makes calls with our softphone (with IAX2). Sometimes the softphon freezes. The call is ACTIVE but the user cant hang it up. At this time in the log file (asterisk/messages) appear the next line: channel.c: Avoided initial deadlock. we use: SVN-branch-1.2-r46176M with VoIP channel (ADSL) Can you help me? What is the problem? If you can send us either the output of iax2 debug or an ethereal trace of the packets in a conversation that fails I'll take a look. At a guess your softphone has a bug, and asterisk is just issuing a warning, but I don't have enough evidence yet. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez wrote: Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13 What exactly is the point is such settings? Why not connect directly to the provider over SIP? Or to the ATA over SIP? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Live creation of trunk groups
On Mon, Oct 30, 2006 at 03:08:39PM -0500, Andre Courchesne - Consultant wrote: Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Edit zapata.conf and from the asterisk cli run 'reload' or 'reload chan_zap.so' . This will apply most changes from apata.conf. Basically anything that doesn't change the very nature of the channel. Tat is: you will not be able to create and destory channels that way, or even change their signalling. But you'll be able to change probably all other parameters. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] simultaneous ring - call groups or queues or something else?
Brian Rogan wrote: You can just seperate multiple phones with in the Dial command, as the voip-info wiki page shows: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Thanks! It's not always clear where to look first for these things. I'm repeatedly blown away by the ease of configuration and flexibility of Asterisk. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cheapest way to determine channels in a group from outside asterisk?
On 26 Oct 2006, at 12:12, Nick Adams wrote: I need to determine the number of active calls in a group from outside of Asterisk. Currently I poll the manager API and parse the channel status list but this is becoming too expensive on CPU. What are my options? What is considered standard practice ? Update a DB field? Poll the manager api? Use an asterisk -rv 'some command' call? That depends on your configuration. If you already use SNMP in your organisation, you might want to use that. If you are/have a java coder, there is some support for the asterisk MIB in the free-ware from snmp.westhawk.co.uk (Disclaimer - I wrote large chunks of it so I'm biased :-) ) Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPv6
On 26 Oct 2006, at 15:33, David Bandel wrote: Folks, Anyone know if Asterisk supports IPv6? If not, is support planned? There was a talk at astricon on this. (I think the slides will be available from astricon.net). The short answer is no, not yet, but folks are working on it. Tim. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXO Card's vs. T1
Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk architecure
jez . a écrit : Dear all, I've recently installed Asterisk and am trying to understand where exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work as a proxy. I am specifically interested in SIP. Could anyone perhaps point me out to a diagram with SIP users and Asterisk to better understand how I should set up my network? Thank you Hi, You can find some interesting diagram here : http://www.tech-invite.com/Ti-sip-dialog.html Other diagrams more architecture ortiented : http://lehmann.free.fr/divers/SIP%20tutorial.pdf slides 32 and after. The document is not mine :) If you want something more specific to Asterisk's architecture, I recommand you this book : http://www.eyrolles.com/Informatique/Livre/9780596009625/livre-asterisk.php Bye Guillaume Lehmann ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] overlap of zap trunk groups
Damon Estep wrote: Can anyone suggest any reasons why a zap (PRI) b channel should not be a member of multiple zap trunk group definitions? For example; Group 1 = Channels 1 to 23 Group 2 = channels 1 to 12 Group 3 = channels 13 to 23 The purpose is to restrict the number of channels a particular extensions can use, but use the entire span for other extensions. Part of a production /etc/asterisk/zaptel.conf: group=1 channel = 1-6 group=1,2 channel = 7-12 group=0 channel = 13-16 So the correct solution is to define the channel only once, but the group= parameter can contain many groups delimited by a comma, correct? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DTMF Tones
Jason Walker wrote: I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct DTMF tones 25% of the time. I have to call several times to enter an extension. I have a router and a packet shaper and some other stuff. Anyone have any other ideas why this might happen. I do not have any Zap channels but I am running CentOS4. I also do not have any cards installed. Thanks What phones and codec are you using? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Characters in CLI on TTY9
On Tue, Oct 31, 2006 at 02:03:33PM -0500, Forrest Beck wrote: When I look at TTY9 (using init.d and safe_asterisk to start the asterisk process), I am getting some strange characters. When a application is run the and the CLI shows the application executing the languange almost looks russian...?? Anyone seen this before? http://picasaweb.google.com/jonforrest.beck/AsteriskCLI Bogus terminal settings show color as cyrillic. vim with syntax hilighting will probably give you a similar result. Consult your distro's gurus. Some relevant keyfors: consolechars , setfont -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom or Cisco Phones?
On 10/31/06, Joao Pereira [EMAIL PROTECTED] wrote: Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira I have a Cisco 7960 here in the home office, I recenty purchased a Snom 300 for the lounge. I wrote a very quick mini-review on my blog:- http://www.g6phf.co.uk/site/2006/10/05/snom-300-voip-phone-mini-review/ Christian @ Snom, whilst I have your 'ear' here :) Please can you add a backlight to future revisions of the Snom 300, it would be most welcome!! thanks Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Recommendations
On Tue, 2006-10-31 at 13:52 -0500, Carlos Rojas wrote: Hello, I'm working with supermicro servers, for the irq problems with Dell, any people have problems I second the supermicro servers - particularly the opteron range based on Serverworks HS1000 chipset. Excellent stuff. Well designed, no irq problems and no timing problems. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [asterisk-users] Snom or Cisco Phones?
That and Cisco won't give you the time of day if you don't use their stuff ;) We have about 1600 of the Cisco's on campus, and unless you run them on the call manager, you're not gonna have nearly as many features as any other phone that's designed with SIP in mind. That said, if you need a phone with dialtone, a pretty screen, and limited xml services, then I will say that the cisco's are extremely easy to provision once you figure out the upgrade paths. (Oh, and we're running 7940's and 7960's... if you're looking at the 7912's, etc, good luck, they're a _complete_ pain to work with) Aaron On Tue, 2006-10-31 at 20:41 +0100, Christian Stredicke wrote: I think one of the differences is: We do pay attention to Asterisk and this mailing list ;-) CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira Gesendet: Dienstag, 31. Oktober 2006 13:47 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Snom or Cisco Phones? Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom or Cisco Phones?
On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote: Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia cars Not really. Both are very good phones. * My Clients prefer cisco because it looks more business-like. - The new snom phones do look better though and the side car rules. * The Cisco phone 'feels' very good in your hand, and the voicequality is superb. (I'd say slightly better than that of the snom 360) * Technically, I find the snom phone more advanced and I can do more cool stuff with it - Cisco doesn't seem to like giving features away in SIP. * Snom phones, for example, have freely programmable buttons that can park/retrieve/transfer calls, show line status etc. I can't get that to work with Cisco phones at all. * Putting custom ringtones (and choosing which ones to use) is a no-brainer with snoms and real trouble with ciscos. * On ciscos, I find the upgrade path from sccp to sip a totally unnecessary annoyance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] compilation problem with asterisk-addons
Hi, Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this: Note: MySQL libraries are installed and the structure is as follows: /usr/src/astsources/asterisk-1.2.13 /usr/src/astsources/asterisk-addons-1.2.5 in /usr/src/astsources/asterisk-addons-1.2.5 I do: make clean make and the output is: ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:25:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:26:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:27:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:28:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:29:29: asterisk/module.h: No such file or directory app_addon_sql_mysql.c:30:34: asterisk/linkedlists.h: No such file or directory app_addon_sql_mysql.c:31:31: asterisk/chanvars.h: No such file or directory app_addon_sql_mysql.c:32:27: asterisk/lock.h: No such file or directory app_saycountpl.c:11:27: asterisk/file.h: No such file or directory app_saycountpl.c:12:29: asterisk/logger.h: No such file or directory app_saycountpl.c:13:30: asterisk/channel.h: No such file or directory app_saycountpl.c:14:26: asterisk/pbx.h: No such file or directory app_saycountpl.c:15:29: asterisk/module.h: No such file or directory app_saycountpl.c:16:27: asterisk/lock.h: No such file or directory cdr_addon_mysql.c:23:29: asterisk/config.h: No such file or directory cdr_addon_mysql.c:24:30: asterisk/options.h: No such file or directory cdr_addon_mysql.c:25:30: asterisk/channel.h: No such file or directory cdr_addon_mysql.c:26:26: asterisk/cdr.h: No such file or directory cdr_addon_mysql.c:27:29: asterisk/module.h: No such file or directory cdr_addon_mysql.c:28:29: asterisk/logger.h: No such file or directory cdr_addon_mysql.c:29:26: asterisk/cli.h: No such file or directory res_config_mysql.c:41:30: asterisk/channel.h: No such file or directory res_config_mysql.c:42:29: asterisk/logger.h: No such file or directory res_config_mysql.c:43:29: asterisk/config.h: No such file or directory res_config_mysql.c:44:29: asterisk/module.h: No such file or directory res_config_mysql.c:45:27: asterisk/lock.h: No such file or directory res_config_mysql.c:46:30: asterisk/options.h: No such file or directory res_config_mysql.c:47:26: asterisk/cli.h: No such file or directory res_config_mysql.c:48:28: asterisk/utils.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/astsources/asterisk-addons-1.2.5/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c common.c:1:29: asterisk/logger.h: No such file or directory common.c: In function `decode_header': common.c:93: warning: implicit declaration of function `ast_log' common.c:93: error: `LOG_WARNING' undeclared (first use in this function) common.c:93: error: (Each undeclared identifier is reported only once common.c:93: error: for each function it appears in.) make[1]: *** [common.o] Error 1 make[1]: Leaving directory `/usr/src/astsources/asterisk-addons-1.2.5/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 Thanks for your help. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time
On 10/31/06, Brad Templeton [EMAIL PROTECTED] wrote: On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote: Have you tried setting the externalip and localnet parameters? Localnet makes some sense, and is set (should be the default anyway, no?) externalip, as I understand it, is for an Asterisk which is behind a NAT. This asterisk is not behind a NAT to anybody. The phones are behind a NAT to the outside world but not to the Asterisk box, which has two ethernets on it, one for the internal natwork and one for the real internet. It uses bindaddr=0.0.0.0 and listens to both addresses. Sorry for my previous post I misunderstood the problem. You should set canreinvite=no to all sip peers that connect from outside. That's precisely what I don't want to do. This would block native bridging in the one case where it's most important. The correct behaviour, as I see it is: a) Native bridge when connecting two external channels -- everybody is on the real internet It might not work if one of them is NATed. Therefore the correct way to do this is to use canreinvite=no b) Native bridge when connecting two internal channels -- everybody is on the 192.168.* network canreinvite=yes will take care of this. c) Route RTP through Asterisk when connecting internal and external Again by adding canreinvite=no to externals you have this. d) When a channel is to a device behind a remote NAT, the usual rules apply (either use STUN or other smart NAT, or route RTP through Asterisk) How will asterisk know? The correct *setting* (not behavior) is canreinvite=no for the external devices. The super correct behaviour, which I don't expect but would be nice is e) Clever native bridge between internal and external by being aware that the device talks to the outside world using a different address than it talks to you. (Possibly if the phones use STUN they will tell Asterisk their external IP, which is not the same as Asterisk's though it's on the same subnet) I have used localnet=192.168.* and nat=yes on a local device and it still attempts an incorrect native bridge between internal and external, with one-way audio. If I do canreinvite=no on the local devices then it works of course, but now means the local phones will never native bridge amongst themselves. In a larger network, that would be a problem, and it's a poor result in any network. Why are you so against having the RTP go thru asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: IAX2 show peers - description
Hi friends, thank you for comments... Marian Tomislav Parčina napsal(a): In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I think the (T) is for Trunk. Regards Fred Hi Fred! I believe that T is for trunk. Thank you. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marian Rychtecky [EMAIL PROTECTED] Tel. +420 724 397 441 ICQ 76582857 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
I forgot to mention that the Carrier that owns the ATA box was not willing to let me connect directly over IP, I was only allowed to use the FXS port. He already ack that he has a problem with disconnections. On 10/31/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez wrote: Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13 What exactly is the point is such settings? Why not connect directly to the provider over SIP? Or to the ATA over SIP? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards vs. Channel bank with T1
Looking at the number's now it seems that a T1 will be more. Anyone here sell PRI's ? - Original Message - From: Jay R. Ashworth [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 31, 2006 9:38 PM Subject: Re: [asterisk-users] FXO Cards vs. Channel bank with T1 On Tue, Oct 31, 2006 at 08:20:57PM +0200, Dovid B wrote: Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? If you need enough ports to make a T-1 card cost-efficient, then you might oughtta be looking at an Ethernet to FXO media gateway instead -- assuming you need analog interfaces. FXO side, why not just go T-1 or PRI? Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] overlap of zap trunk groups
Damon Estep wrote: Damon Estep wrote: Can anyone suggest any reasons why a zap (PRI) b channel should not be a member of multiple zap trunk group definitions? For example; Group 1 = Channels 1 to 23 Group 2 = channels 1 to 12 Group 3 = channels 13 to 23 The purpose is to restrict the number of channels a particular extensions can use, but use the entire span for other extensions. Part of a production /etc/asterisk/zaptel.conf: group=1 channel = 1-6 group=1,2 channel = 7-12 group=0 channel = 13-16 So the correct solution is to define the channel only once, but the group= parameter can contain many groups delimited by a comma, correct? Correct. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compatability
Hi All, I have a new client who has an existing Asterisk PABX and is looking for us to install a TE110P for him, However he has a Dell SC420 and I have never used one before. I have had no problems with any other Dell servers which we use almost exclusively. Has anyone had any good/bad experiences with the SC420 in relation with Digium cards? Thanks for your help. Joel Asterisk IT www.asteriskit.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Call Statistics
Check out voip-info.org, there are quite a few GUIS some even generate nice graphs!On 10/31/06, omar parihuana [EMAIL PROTECTED] wrote:Hi Folks,I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format,arethere someopen source aplication for Asterisk that be easier for use. Plsanything suggestion will be very appreciate.ThanksRgds.-- Omar E.P.T-Certified Networking Professionals make better Connections!http://omarept.blogspot.com/Usysnet CorpOpen Source Solutions www.usysnet.com.pe___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones
Where are these DTMF tones going? From where? Be specific, post the relevant config file sections I can't read minds and I'd be surprised if 0.1% of the people who read this can either On 10/31/06, Jason Walker [EMAIL PROTECTED] wrote: I have tried beta2, beta3 and now back to 1.2.12.1 and I have correctDTMF tones 25% of the time.I have to call several times to enter anextension.I have a router and a packet shaper and some other stuff. Anyone have any other ideas why this might happen.I do not have anyZap channels but I am running CentOS4. I also do not have any cardsinstalled. Thanks___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Server Recommendations
I'm working with Supermicro as well. -Original Message- From: Carlos Rojas [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 31, 2006 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Server Recommendations Hello, I'm working with supermicro servers, for the irq problems with Dell, any people have problems Regards On 10/30/06, Paul Hales [EMAIL PROTECTED] wrote: How many analog lines are you looking at? Hundreds? PaulH On Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote: We have a number of clients who will be needing a server to host Asterisk on. Many of these clients use analog (FXO) lines that will need to be connected to Asterisk via Sangoma cards. Can anyone recommend an industry-standard server (like IBM, Dell, HP, etc.) that has enough open PCI slots to handle up to six of the Sangoma cards? We would like to be able to tell the customer to just go purchase this model server from this manufacturer and it will work. Suggestions? Thank you! Joe Dennick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [OT] wi-fi ip phone scenario
I've done extensive testing, WDS is just as reliable as wired,however at first we had issues with some AP that would not respond and needed to be rebooted. But if its possible to wire the AP you should since WDS will eat alot of bandwidth and also decrease the range since most the AP will have to be within range of eachother, way more than overlapping coverage Alberto:I would suggest you try to keep all the AP on the same channel. With that large of a space I wouldnt expect too much interferance from the outside.On 10/28/06, Martin Joseph [EMAIL PROTECTED] wrote: On 2006-10-27 11:55:14 -0700, Andrew Joakimsen [EMAIL PROTECTED] said: Are you using WDS? While it won't totally fix every issue, I've found in my trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but re-registartion won't require a reboot).I think it's cleary true that wiring WIFI infrastructure is easier andmore reliable then WDS.On the other hand,I have been running my little network with WDS for over three weeks now, and it has been completely reliable.The tricks where to configure things properly and to have the basescloser together then one would think would be needed.Once this was setup. It works, and it keeps working.We had a couple of stress tests also, one black out and one unplugged router(carpenter).Came up cleanly and continued working fine.No mis-registrations andno problems.Marty___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Panasonic KX Model
I did this today with a Panasonic KX-TD1232 and a Digium TDM2401E Card. I hope to put it on the wiki soon, if you need help just tell me with what. On 10/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: If Someone did that, How I connect extensions.conf with this type of Hybrid system to work with asterisk inside this schema: PSTN---PANASONIC KX -- Asterisk | |-send internal call Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users