Re: [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing
exten = 9,1,NoOp(ISDN: Pickup outside line (early B3 connect) for: ${CALLERIDNUM})
exten = 9,2,SetCallerId(${THORCOM_MAIN}) exten = 9,3,Dial(CAPI/g1//b) exten = 9,4,Hangup
use this string with BT
extn = 9,3,Dial(CAPI/g1//bo)
Should provide correct progress
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Re: [Asterisk-Users] Asterisk on PPC chan_capi issue
chan_capi registers fine: ** [chan_capi.so] = (Common ISDN API for Asterisk) == This box has 1 capi controller(s). == Reading config for BRI1 -- ast_capi_pvt BRI1-pseudo-D (MSN1,MSN2,capi-in,0,2) (1,4,128) -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128) -- ast_capi_pvt BRI1 (MSN1,MSN2,capi-in,0,2) (1,4,128) -- listening on contr1 CIPmask = 0x1fff03ff == Registered channel type 'CAPI' (Common ISDN API Driver ($Revision: 1.115 $) ) == Registered application 'capiCommand' == Registered custom function VANITYNUMBER Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2): ** == BRI1: Incoming call 'my GSM' - 'MSN2' -- Executing Macro(CAPI/BRI1/MSN2-0, stdexten|1003|SIP/1003) in new stack -- Executing Dial(CAPI/BRI1/MSN2-0, SIP/1003|10|TtwW) in new stack Dec6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No translator path exists for channel type SIP (native 65535) to 0 Dec6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) Looks like a codec problem when making calls to the SIP phone, ensure your sip phone has Alaw enabled in sip.conf, and supports the g711alaw codec. In its config Jason ___ --Bandwidth 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] sipuras 841 bad sound
Make sure you have turned off VAD as asterisk does not support Silence supperssion. Jason On 9/21/05, Juan Jose Comellas [EMAIL PROTECTED] wrote: Have you tried upgrading the firmware? I had several problems with theoutbound volume of these phones until I upgraded them. On Tuesday 20 September 2005 20:46, Anton Krall wrote: Hi Guys! I have a problems with some sipuras 841 and asterisk 1.0.9. Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with steve's unicall. Everything compiled fine and in fact I can make and receive calls but I have a problem with bad sound when the sipuras call the outside E1's lines. I can listen to the caller without problems but they heard me with a choppy sound as if you were losing frames or cutting off. Calls between internal sipuras sound good (eventhough the speaker and headset sound comes and goes, for example, when you start talking, seems as if the sipuras takes a few seconds to catch up with you on volume so the remote user listen to you as if the first words and the last were at low volume and the conversation in the middle sound good, any had that problem?) So, internal calls sound good between 841's but sound volume is weird at the start and end of a sentence. Calling the outside lines via E1's, I can listen to people without problems but they heard me as choppy or cut off. Anybody had issues like this? Is it asterisk or the phones or what? Hope you can help Guys, Im really banging my head against the wall here. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Juan Jose Comellas([EMAIL PROTECTED])___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does one set-up incoming/outgoing SIP with no registration and only IP authentication?
On 9/19/05, Frank Tarczynski [EMAIL PROTECTED] wrote: I'm new to asterisk and need some help with ideas to handle thisconfiguration question.I am trying to establish a termination point/DID number in another country.I am currently running Asterisk CVS-HEAD.My foreign provideruses SIP and authenticates via IP address.I am not required toregister my SIP connection in order to send or receive calls.[from-200.XXX.XXX.XXX ]type=userhost=dynamicallow=gsmallow=ulawnat=yescanreinvite=yescontext=outgoinginsecure=very Change the host line to host=200.XXX.XX.XXX as you are not registering with the host ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange problem with G711/G729, Cisco and Grandstream
But when BT-100 calls 7960 the following is happening: -- Executing Dial(SIP/3710-8f2b, SIP/1707|15) in new stack -- Called 1707 -- SIP/1707-e96a is ringing -- SIP/1707-e96a answered SIP/3710-8f2b -- Attempting native bridge of SIP/3710-8f2b and SIP/1707-e96a May 4 16:46:58 WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot native bridge. sipsrv1*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg 192.168.128.171 170702fff7f7169 00102/0 ulawTx: ACK 67.126.23.2513710b5d3f977ea1 00101/52181 g729Rx: ACK When this bug is gonna be fixed? Change the codec order in the phone configuration and place g729 higher it is not asterisk doing this ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] music on hold on R key not working.
On 4/28/05, Eugenio De Vena [EMAIL PROTECTED] wrote: Oh boy I am getting crazy... I installed an asterisk with J4BRI ( 3 BRI point to point ) , Snom phones and everything works fine. Where's the problem? Well I can not get music on hold.. Well really MusicOnHold works, works on Queue, works on # The release is Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k 1.0.6 had a broken hold music you need 1.0.7. and then bristuff it ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap/PRI: received AOC-E charging
On 4/26/05, Matthew Boehm [EMAIL PROTECTED] wrote: Trying to make a call via our PRI: (CVS everything, CVS-NHEAD-04/23/05-16:08:12) -- Executing Dial(IAX2/[EMAIL PROTECTED], Zap/R2/2815699900|30) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called R2/2815699900 -- Channel 0/19, span 2 got hangup -- Channel 0/19, span 2 received AOC-E charging 0 units Apr 26 09:06:49 WARNING[10040]: chan_zap.c:7457 zt_pri_error: PRI: Call From Your debug Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) ] Looks like either a number problem or no route to destination. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over solutions
On 4/26/05, snacktime [EMAIL PROTECTED] wrote: On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote: Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? That's kind of a loaded question... Do you plan on expanding? What is your budget? What are your uptime requirements? Are you serving customers or is this just for internal use? The biggest problem with that solution is voicemail it could get left on one server and not be on the other for one hour. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 earpiece speaker echo question
On 4/26/05, Jeremy Koski [EMAIL PROTECTED] wrote: Normally, when you speak into the receiver of a phone, you can hear yourself in the earpiece at a very low volume. I have a Cisco 7960 phone that I'm using with asterisk and I don't get that echo back on the earpiece speaker. I only have one Cisco 7960 phone, so I can't test it on others right now. My question is...Is this normal, do I have a bad handset? Is a way I can fix it? On my 7960 if I blow accross the mouthpiece I can hear it quietly in the earpiece (at least when dialtone is heard) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On 4/27/05, Andre Normandin [EMAIL PROTECTED] wrote: Does anyone know what the [WARNING: . Changethread: Can't change device '**Unknown**'] line means below.. I just set verbosity to level 5, and noticed that error everytime a voicemail is left.. Everything seems to work ok, and I have no idea how long that error has been there, but I'm just curious if it is something important :-) Looks like the call is coming out of voicemail and then going somewhere else or you have an exten _. defined that is catching a hangup, post your extensions.conf for further analysis. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Gateways Asterisk
On 4/26/05, Callum McGillivray [EMAIL PROTECTED] wrote: My problem is that this installation is most likely to occur prior to the release of the new card (and definitely prior to it's vigorous testing in the field). If anyone can give me ideas at this point it would be appreciated. There are a number of cisco routers that will do the job for you but they are not cheap eg AS5350 8 E1's AS5400 16 E1's or other normal routers such as 3660's will support upto 4 E1'e Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan capi: Long incomingmsn line in capi.conf?
On 4/26/05, Stefan Gofferje [EMAIL PROTECTED] wrote: Stefan Helbing schrieb: Hello, the incomingmsn line in chan_capi's capi.conf is limited to 80 characters (AST_MAX_EXTENSION default value). My problem: I have to include several MSNs but NOT all. The interface is a 30 channel PRI card with a number area of 600 numbers, splitted in different functions. Some numbers are used for fax, some for PPP, some for telephony. According to another email on this list, accept all incoming MSN's but create an entry in extensions.conf for each msn you wish to ignore (or wildcard) as follows exten = _123456XX,1,Wait(30) The wait will stop asterisk from answering the call so the other capi devices fax etc should then answer the call. Try it and let us know it would be good for future reference Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP/NetMeeting
On 4/26/05, William C. Lohr Jr. [EMAIL PROTECTED] wrote: Does anyone know if it is possible to resolve an IP from outside a small LAN. I would like to be able to specify a SIP client that is outside my office LAN. The problem is that the isp will not provide a static IP that's affordable. I use a DYNDNS.org address with it. When I want to use NetMeeting for desktop sharing, I just ping the DYNDNS address and it gives me the current IP of the remote machine. Is it possible to specify the host name, say billscomputer.dyndns.org for the address of the SIP client in the appropriate .conf file for Asterisk? This is covered automatically if you set host=dynamic in sip.conf and have the sip phone register with your asterisk then asterisk knows what IP address the phone is on, this will be updated with every registration request. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium for ETSI ISDN
On 4/26/05, Nathaniel Angelo A. Torres (247talk) [EMAIL PROTECTED] wrote: Hi, I just wanted to know if Digium support ETSI ISDN? Yes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Gateways Asterisk
On 4/26/05, Michael Baird [EMAIL PROTECTED] wrote: You might look into the Lucent TNT's, they do SIP/MGCP (with the Hash codes, os 10.1.xx+), and also terminate modem calls. They are cheap (check ebay, www.qualitek.net) and their are loads of them out there. One TNT will handle your requirements easily, their is an example on the wiki on how to use a TNT with asterisk, and it works properly. I used Asterisk to talk to them via SIP, didn't try mgcp but it should work fine. The cisco routers will do sip as well. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi: no dialstatus, no causes, no branches
On 4/22/05, Stefan Gofferje [EMAIL PROTECTED] wrote:
Hi folks,
I'm using a Fritz!PCI with chan_capi 0.3.5.
I found that chan_capi neither seems to signal Busy or Congestion to
callers from ISDN nor does it seem to set HANGUPCAUSE, CAUSECODE or
DIALSTATUS if an outgoing call fails. There is also no branch to n+101
if the called party is busy.
Are there any known solutions how to get this working?
A simple one is to change your dial string from
Dial(CAPI/MSN:${EXTEN}) to Dial(CAPI/MSN:b${EXTEN}) This will provide
busy tone from the Carrier
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Re: [Asterisk-Users] Where to post my impovements to ASTCC?
On 4/3/05, Ronald Wiplinger [EMAIL PROTECTED] wrote: You can't see the sweat, but ... I would like tp post my improvements to ASTCC somewhere, ... but where??? Post them as patches to bugs.digium.com and then they can be incorperated into the main code. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi segfault when incoming call is answered
On 4/7/05, Thomas Andrews [EMAIL PROTECTED] wrote: On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote: I have a Fritz! card set up to use capi, however when incoming calls to the card are answered, asterisk segfaults. Have you tried a make clean then make install in the chan_capi source directory make sure the header files are built correctly. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS-HEAD and CheckGroup/SetGroup
On 4/20/05, Sean A. Newton [EMAIL PROTECTED] wrote:
Do the SetGroup and CheckGroup functions behavior differently in CVS-HEAD
vs CVS v1-0?
When I upgrade to CVS-HEAD my call waiting disable doesn't seem to work,
using:
exten = s,1,SetGroup(SIP${ARG1})
exten = s,2,CheckGroup(1)
exten = s,3,Dial(Sip/${ARG1},15,t)
Do you not need a
exten = s,103,Congestion()
otherwise the checkgroup has nowhere to go ?
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Re: [Asterisk-Users] chan_capi looking for missing channel_pvt.h
On Mar 31, 2005 3:32 PM, Mimmus [EMAIL PROTECTED] wrote: Hi, I'm trying to compile channel_capi with current Asterisk CVS. Asterisk compiled successfully but channel_capi (patched with all patches needed, as suggested from some nice people on IRC #Asterisk) compilation fails with: app_capiFax.c:34:34: asterisk/channel_pvt.h: No such file or directory I haven't such file on my system! Peraphs patches are for older CVS versions? Look in the Makefile for a reference to app_capiFax and remove it. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music Answer while waiting
On Mar 31, 2005 1:00 PM, Robson Ribeiro [EMAIL PROTECTED] wrote: Hi, If I want a user to, while waiting for a transfer after responding to an IVR, to listen to music instead of a ring sound, what is the change should i do in extensions.conf? Is it on the IVR menu or on the optional extension The change id one in the dial command that calls the extension show application dial in the cli will help look at the m option ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reject second IAX call
On Mar 31, 2005 12:31 PM, Marc SCHAEFER [EMAIL PROTECTED] wrote: Hi, is there a configuration in iax.conf to specify that if a call goes to that peer, a second call should not be allowed. Specifically, I do this: Dial(IAX2/iaxcomm) # in extensions.conf for a specific extension in iax.conf: [iaxcomm] type=friend mailbox=20 accountcode=iaxcomm username=iaxcomm host=dynamic auth=md5,plaintext,rsa secret=fksjdfh73 ; changed context=local-iaxcomm permit=192.168.10.0/24 allow=ulaw is there an option to disable a 2nd call? thank you. look on wiki for set group and check group this can do what you need ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setting SIP to dial PSTN with TDM400P
On Mar 31, 2005 1:05 PM, Muhammad Haris [EMAIL PROTECTED] wrote:
I've setup * with TDM400P w/1 FXS, 1 FXO modules.
I've one analog phone connected to TDM400P FXS module, 1 PSTN line to
one of the FXO module(ZAP) , and IP phone connected to asterisk on
LAN.
The calls between SIPs and zap phone (fxs) are OK. But 2 issues
cannot be solved:
1. To dial to PSTN via zap phone, the setup in extensions.conf with
the following
exten = _Nxx, 1, zap/1
doesn't work.
your line is not correct try this
exten = _NXX,1,Dial(Zap/1/${EXTEN})
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Re: [Asterisk-Users] OT: does Sipura SPA 3000 support UK caller id?
On Tue, 22 Mar 2005 10:45:42 -0800, Trevor Peirce [EMAIL PROTECTED] wrote: Mike Dent wrote: Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? Yes it does Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help Debugging my code?
do you really have
[specialized]
[specialized]
it is twice try removing one entry
Jason
On Wed, 23 Mar 2005 02:37:42 +, Scheda [EMAIL PROTECTED] wrote:
Hey, I'm currently using the GotoIf application to set it so if
certain caller ID's call my number, it will transfer it to my cell
phone, here is the code I have so far. I get an error message that
states call rejected by 198.22.67.70: No such context/extention.
when I call the number from my house number. Anyway, here is the code
I have.
[inbound]
exten = 8667393960,1,Answer()
exten = 8667393960,2,GotoIf($[${CALLERIDNUM} =
${house}]?specialized,8667393960,1:2)
exten = 8667393960,3,GotoIf($[${CALLERIDNUM} =
${kendra}]?specialized,8667393960,1:2)
exten = 8667393960,4,GotoIf($[${CALLERIDNUM} =
${rob}]?specialized,8667393960,1:2)
exten = 8667393960,5,GotoIf($[${CALLERIDNUM} =
${jen}]?specialized,8667393960,1:2)
exten = 8667393960,6,GotoIf($[${CALLERIDNUM} =
${mom}]?specialized,8667393960,1:2)
exten = 8667393960,7,GotoIf($[${CALLERIDNUM} =
${dad}]?specialized,8667393960,1:2)
exten = 8667393960,8,Wait(3)
exten = 8667393960,9,Background(/root/asterisk-1.0.6/sounds/ast-intro)
exten = 8667393960,10,Wait(12)
exten = 8667393960,11,Hangup()
[specialized]
[specialized]
exten = 8667393960,1,SetCallerID(${cid})
exten = 8667393960,2,Wait(1)
exten = 8667393960,3,SetMusicOnHold(danecook)
exten = 8667393960,4,Dial(${TRUNK}/${scheda},35,t)
exten = 8667393960,5,Hangup()
I have all the global variables set up correctly, so I'm not sure what
it is exactly
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Re: [Asterisk-Users] Ext matching problems
On Mon, 21 Mar 2005 15:03:14 -0400, Francisco Moreno [EMAIL PROTECTED] wrote: Now, when I dial from any of the ext. to '0' It actually matches the '0', plays the goodbye message, but doesn't hangup but gets directly to the 'pasvalide' context. Same thing happens when I dial to the ext. 1002 (the one that doesn't have voicemail), either it rings further than 10secs or it's busy, it does not hangup but gets straight to the 'pasvalide' context. As far as I understood, it should not happen, it should go through the dialplan leaving those context included at the end and in the orther they are included. your pavalide context is the problem [pasvalide] exten = _.,1,Answer() exten = _.,2,Playback(invalid) exten = _.,3,Playback(goodbye) exten = _.,4,Hangup() _. matches all numbers including h which means hangup, change pasvalide to this [pasvalide] exten = _X.,1,Answer() exten = _X.,2,Playback(invalid) exten = _X.,3,Playback(goodbye) exten = _X.,4,Hangup() And all should be good Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fun with CAPI
On Thu, 24 Mar 2005 14:19:20 +, Gavin Hamill [EMAIL PROTECTED] wrote: Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension numbers.. get a dialtone and then dial onward from there... use show application disa in the cli and send them there rather than playing dial tone, this should do what you want Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Group Ring after Timeout
On Fri, 18 Mar 2005 13:57:37 +0100, Reuben Grech [EMAIL PROTECTED] wrote: Dear All, Would like to know what I should do to:: pickup call immediately and simultaneously Ring a Group, so that caller is listening to message whilst group phones are ringing and first one to pickup gets the call. The dial command will call more than one device eg: exten = 1234,1,Dial(SIP/1234SIP/2345) extension 1234 will now ring sip device 1234 and 2345 Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Group Ring after Timeout
On Fri, 18 Mar 2005 16:59:36 +0100, Reuben Grech [EMAIL PROTECTED] wrote: Dear All, I am listening to blips during conversations when I have an incoming call from an X100P card. This does not happen on all conversations. Any clues? :) Check cat /proc/interrupts make sure the X100P has it's own interrupt I suspect not. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extension.conf dialplan
On Thu, 17 Mar 2005 02:42:27 -0800 (PST), Kamran Ahmad [EMAIL PROTECTED]
wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk is dialing
Executing Dial(OH323/R11429, OH323/40923335224005)
but i want him to dial
Executing Dial(OH323/R11429, OH323/923335224005)
You need to change your dial comand to this
exten = _40,1,Dial(OH323/${EXTEN:2})
the :2 deletes the first 2 digits and removes the leading 40
Jason
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Re: [Asterisk-Users] Asterisk retains DTMF Control Even whenan External IVR System is dialed
On Tue, 15 Mar 2005 14:13:40 -0500, Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote: atxfer = *2 ; Attended transfer Remove attended transfer capability and then you will be able o enter *2XXX Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
On Sun, 13 Mar 2005 21:49:52 +0200, Dimitris Kounalakis [EMAIL PROTECTED] wrote: Hello *Martijn, Thank you for your response. *That was my opinion too, it looses the context due to a bug, and can anyone confirm it also? But I have no output from the command Show channels, and it happens so quickly that it is impossible to issue the command before falling to the default context. In the logs, I can see that the channel exists like CAPI[contr1/2810211694]/0 but this is druring call only. Any other way to debug it more (or to solve it)? My /etc/asterisk/capi.conf is: - [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] controller=1 msn=2810111694 incomingmsn=* devices=2 softdtmf=1 callgroup=1 context=isdn On my system I have the devices=2 as the last line this works for me [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=330417 incomingmsn=* controller=1 softdtmf=1 accountcode= context=from-pstn echocancel=yes echotail=64 devices=2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.0.6 music on hold bug ?!
On Fri, 11 Mar 2005 17:45:55 +0200, Calin Serbanescu [EMAIL PROTECTED] wrote: hello list, last night i upgraded my asterisk box from 1.0.5 to 1.0.6 and my music on hold did not work anymore. Download version 1.0.7 from Cvs this has the fixes in it ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960
Modify the dialplan.xml on your tftp server to this DIALTEMPLATE TEMPLATE MATCH=* Timeout=4 User=Phone/ /DIALTEMPLATE Jason On Thu, 10 Mar 2005 10:31:34 -, Marshall, Ed [EMAIL PROTECTED] wrote: Hi There I am currently having an issue with a Cisco 7960. The phone is using SIP firmware version 6.3. I have successfully got the phone to register with Asterisk and I can call the phone from other non Cisco handsets. However when I dial out from the 7960 I do not even see any output on the Asterisk console. Is there some sort of DTMF setting which I might have incorrectly set ? Following DTMF settings are in my SIPdefault.cnf file. dtmf_inband: 1 dtmf_outband: avt dtmf_db_level: 3 It seems to me like Asterisk is not detecting any key tones from the phone. I have followed a number of setup guides for this phone to no avail. Any help or suggestions are greatly appreciated. Regards Ed ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-oh323-0.7.1 compile error
try with a CVS head before 03/03/05 as the channel structure was changed then, or get an updated version of asterisk-oh323 if there is one availiable Jason On Thu, 10 Mar 2005 06:25:04 +0330, mohammad [EMAIL PROTECTED] wrote: Hi; I use the following asterisk, openh323, pwlib: asterisk = cvs-head-03/09/05 openh323 = 1.13.5 pwlib = 1.6.6 asterisk-oh323= 0.7.1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hide callerid via presention bits of ISDN
On Thu, 10 Mar 2005 16:22:39 +0100, Deti Fliegl [EMAIL PROTECTED] wrote: Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? look at this extension.conf command show application SetCallerPres asterisk*CLI -= Info about application 'SetCallerPres' =- [Synopsis]: Set CallerID Presentation [Description]: SetCallerPres(presentation): Set Caller*ID presentation on a call. Always returns 0. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cicso 7912 phones 3 out of 8 not grabbing thegkMAC file
On Tue, 08 Mar 2005 17:34:31 -0500, Jerry Geis [EMAIL PROTECTED] wrote: SEPDEFAULT.cnf is not required nor recommended. The 7912 only uses the gkMAC file and the software version CP7912XXX file The gk file must be lower case.. This phone 192.168.255.250 is requesting SEPXXX It is still running a SCCP Imagege not SIP and needs Upgrading ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial option g
Could you do something with the h (Calling party Hangup)
eg
exten = h,1,DoSomething
On Sun, 6 Mar 2005 15:00:23 -0500, George Burt [EMAIL PROTECTED] wrote:
I am trying to run a macro at the beginning of call and after the call is
terminated.
exten = 33,1,Macro(makeOnJS,${EXTEN},${CALLERIDNUM},${DATETIME})
exten = 33,2,Dial(SIP/33,15,tg)
exten = 33,3,NoOp(makeOffJS*${EXTEN}*${CALLERIDNUM}*${DATETIME})
exten = 33,4,Macro(makeOffJS,${EXTEN},${CALLERIDNUM},${DATETIME})
exten = 33,102,Voicemail2(b33) ; go to Voicemail2 if phone is Busy
exten = 33,103,Macro(makeOffJS,${EXTEN},${CALLERIDNUM},${DATETIME})
exten = 33,104,Hangup ; and then hangup.
This runs the [macro-makeOnJS] just fine.
It runs the [macro-makeOffJS] only when the called party hangs up first.
In fact, that is exactly what the option g description says in the Dial
documentation:
g: When the called party hangs up, exit to execute more commands in the
current context.
In the Return Codes description of the Dial Command, it says:
Dial returns -1 if the originating channel hangs up, or if the call is
bridged and either of the parties in the bridge terminate the call.
I need a way to do something if the Dial returns a -1 code.
Any ideas?
Thanks,
George Burt
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Re: [Asterisk-Users] TDM22B in the UK on BT
On Tue, 08 Mar 2005 13:04:48 +, Dan Goscomb [EMAIL PROTECTED] wrote: Hi I am having problems getting my card to hang up properly when a remote party hangs up the line. With BT you do not need to use, Busy detect the power inversions will disconnect for you however when the far end clears a call you originated then the exchange you are connected to will hold up the call for 20-30 seconds before disconnecting (this allows the called party to place the phone on hook go to another phone and pickup the call and resume the conversation. You will have no control over this. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attended Transfer (ATXFER) with CVS asterisk r 1_
Patch your chan_capi with this and you will be able to compile CVS HEAD http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 Jason On Thu, 03 Mar 2005 18:13:19 +0100, Massimo [EMAIL PROTECTED] wrote: Hi, I successfully installed asterisk 1.0 with Capi 0.35. In my pbx system I would like to use the atxfer function but is not included in the stable asterisk. Is there a way to include it in my version of asterisk: I did no used the last cvs because I can't compile the chan_capi .in it. :( Bye ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi with patch compilation error
The HEAD version was changed last night to be incompatible with the patch I provided, My C skills are not good enough to fix this so you need to checkout from cvs yesterdays code cvs checkout -D 03/03/05 asterisk Jason On Fri, 04 Mar 2005 13:51:51 +0100, Massimo [EMAIL PROTECTED] wrote: Hi, I'm trying to make work chan_capi with last asterisk CVS. I installed last zaptel,libpri,last cvs ana patched chan_capi 0.35 with the patch kindly suggested me by Jason Williams: http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 First I received error 127 that I resolved commenting the line CC=gcc-2.95 but now I have this error: chan_capi.c: In function `load_module': chan_capi.c:2843: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_capi.c:2843: too many arguments to function `ast_channel_register' chan_capi.c: In function `unload_module': chan_capi.c:2863: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type make: *** [chan_capi.o] Error 1 Someone can help me ? Bye ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting phpconfig to work?
Try using the url http://ip-of-machine/phpconfig/phpconfig.php On Thu, 3 Mar 2005 12:45:03 +0300 (EAT), Julius Kidubuka [EMAIL PROTECTED] wrote: Hi, I have just tried to get phpconfig to work but to no avail. In my browser I type; http://ip-of-machine/phpconfig/ and this returns the following output; Index of /phpconfig NameLast modified Size Description Parent Directory03-Mar-2005 12:15 - asterisk.reload 03-Mar-2005 12:28 1k cls_phpconfig.php 03-Mar-2005 11:4814k cls_phpconfig_html.php 03-Mar-2005 11:5517k images/ 24-Feb-2005 09:06 - phpconfig.php 14-Sep-2003 19:32 6k phpconfig_init.php 03-Mar-2005 11:44 2k Apache/1.3.33 Server at ip-of-machine Port 80 I have made the necessary changes to all the files in the phpconfig directory and my DocumentRoot is set to /usr/local/www/. To add to this, I have php4-4.3.10_1 and apache+mod_ssl-1.3.33+2.8.22 installed on my Asterisk box. What could I be doing wrong? Thanks in advance! Rgds, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multitenant feature
Contexts can be used to partition Asterisk, but the administration is not multitenanted On Thu, 3 Mar 2005 10:47:03 +0100, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, Has any one tested or know if Asterisk support multitenant PBX, ie the Asterisk support either multiinstances on the same machines or acts exactly as many virtual PBXs to be shared between several small campanies. Thanks for the hint Aref ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling hangup in background
Try adding an exten = h,1,DoSomething in the context Jason On Thu, 03 Mar 2005 14:40:10 +0100, administrator tootai [EMAIL PROTECTED] wrote: Hi everybody, I'm running an IVR menu with different languages setted up by user when they enter this menu. What I want is when they hangup, asterisk sets the default language (aka en) back. I'm wondering which extension is called after a hangup in a background command? BTW my IVR menu is in a goto context. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: SIP Aware Firewall with Asterisk
On Mon, 10 Jan 2005 19:38:23 +, John Middleton [EMAIL PROTECTED] wrote: Not an enterprise level system, but anyone used the www.intertex.se IX66? Yes they work great no nat traversal issues, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FOP Asterisk Manager Login Failed?
On Fri, 03 Dec 2004 18:49:32 -0500, Nick Bachmann [EMAIL PROTECTED] wrote: Noah Miller wrote: I've told lots of people about the Flash Operator Panel before, but I've never actually used it myself. I've got it all set up nicely, but I can't seem to authenticate to the asterisk manager (which is running on the same box). When I set the op_server.pl to give debug messages, it shows that it can reach the asterisk manager, but cannot authenticate: ** Asterisk event received, process block... - Action: Login - Username: user - AuthType: MD5 - Key: 0be2f6f6e39f05a53f5a292517ede3e2 ** End of block - Response: Error - Message: Authentication failed I note that it says the authentication is done with MD5, do I need to put an MD5 hash in for the secret in the configuration files? No. The md5 is used so that your actual secret does not have to be transmitted in plaintext. The concatination of the random key and the secret is computed by both sides and hashed, if these two intermediate forms of your secret are the same you are authenticated. [user] secret = usersecret deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 Is your FOP on a different machine? If so, you'll have to explicitly add its IP or remove the deny statement, as it is blocking all IPs on all subnets. If it is on the same machine also include the IP Address as well as 127.0.0.1 Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fallthrough extension.
On Thu, 2 Dec 2004 22:30:17 +, Jon Lawrence [EMAIL PROTECTED] wrote: Hi all, I'm trying to sort out my dial plan. What I'm wanting is something like the following - a bit simplified but hopefully you'll get the idea. 1) match internal extensions: dial them 2) anything else: send out zap 1 is easy :) it's 2 that's giving me problems. I had hoped that the 'i' extension would act as a catchall extension but it seems to only do that from a menu. I've tried matching _. (hoping that * would parse the dial plan from top to bottom) but that just took over the entire dial plan and everything went out of the dial with the _. match. The right way to do it is two have two contexts see below [internal] exten =_3XXX,1,Dial(. etc include=catchall [catchall] exten = _.,1,Dial(Zap. etc In this way internal will be parsed first then the catchall and followed through in include order. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No version string
On Wed, 1 Dec 2004 18:44:43 -0500, Christopher Jacob [EMAIL PROTECTED] wrote: After it downloads the files, I do a make clean make make install When I connect to the console on one machine I get... Connected to Asterisk CVS-v1-0-10/15/04-14:48:10 currently running on bell ( Try doing a make update then a make install from the asterisk src directory that will update the .version files Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi question
On Sat, 27 Nov 2004 12:17:45 +, Robbie Hughes [EMAIL PROTECTED] wrote: -- started pbx on channel (callgroup=0)! == Starting CAPI[contr1/368466]/33 at isdn,368466,1 failed so falling back to exten 's' -- Called 101 -- Called 102 -- SIP/101-1b74 is ringing -- SIP/102-b2b1 is ringing (extensions.conf) [isdn] exten= s,1,Dial(SIP/101SIP/102,30,tr) exten= s,2,capiCD(020712341234) To fix this warning change your extensions.conf to [isdn] exten= 368466,1,Dial(SIP/101SIP/102,30,tr) exten= 368466,2,capiCD(020712341234) 368466 is the msn that the call is arriving on. Regards Jason PS I think Call Deflection is a service you have to order from BT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a way to check if an extensions exists in a context before you send the call there.
On Tue, 23 Nov 2004 17:14:30 -0700, Chris Modesitt [EMAIL PROTECTED] wrote: Is there a way to check if an extension exists? This is the problem I ran into, I have exceeded the number of extensions you can attempt to match in one pass (1500+ Extensions). The solution is not to fix the extensions.conf parsing but to organise your extensions.conf with wild cards and pattern matching and so reduce the number of extensions. Or app_realtime may solve this in the future Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2-SIP-meetme = ZOMBIE
On Tue, 23 Nov 2004 20:53:57 -0700, Ryan Courtnage [EMAIL PROTECTED] wrote: If someone has both IAX and SIP clients, would you please attempt to duplicate the below problem? I don't want to submit a bug unless the problem can be verified. The SIP client must support attended transfers (ie: sayson, uniden): 1) Make a call from an IAX extension to a SIP extension 2) On the SIP phone, use attended transfer (not #) to transfer the call to a meetme room 3) Execute 'show channels' at the * CLI Do you see any 'zombie channels'? Thanks in advance, Ryan I see the same Shuttle*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Zap/pseudo-400601102 (from-330377 s1 ) Rsrvd (None) (None) IAX2/[EMAIL PROTECTED]/2 (office 8600 1 ) Up MeetMe 8600 SIP/1234-e529 (office 1 ) Up Bridged Call SIP/1234-9cebZOMBIE SIP/1234-9cebZOMBIE (macro-stdexten s7 ) Up Dial SIP/1234|20|tr 4 active channel(s) This is with IAX and a Cisco 7960 following the same steps. The ZOMBIE Remains.. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line load balancing
On Tue, 23 Nov 2004 11:11:02 +1100, Paul Hales [EMAIL PROTECTED] wrote: Also - the mobile phone plans we are using get very expensive after approx 1500 minutes, so we have to make sure that none of the lines go over that! But the zap/r1 options should be OK for a start at least. show application dial is your friend Check out the L option to limit the call to x ms Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEED HELP!!
On Tue, 23 Nov 2004 13:17:57 +, WipeOut [EMAIL PROTECTED] wrote: Please can someone look at my last two posts and try and shed some light onto why my system is dropping calls.. If I don't get it right we will be forced to drop Asterisk which I really don't want to do.. I Have not read you last posts but the usual cause for asterisk dropping calls is the busy detection algorithims, Try turning off busy detect or increasing the busy count variables, in zapata.conf Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Analog Intercom) PagePal by ATT -- was hooked to a Merlin
On Thu, 18 Nov 2004 18:52:21 -0600, Jeb Campbell [EMAIL PROTECTED] wrote: I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of an analog phone to page system, but of course I would like to reuse what is there. Thanks for any advice or pointers, Try this http://www.pagepac.com/pdfs/pagepal.pdf You should be able to connect the system to an analoge port Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incompatible with our capability 0x400.
On Sun, 21 Nov 2004 18:54:25 +0500, khurram bhatti [EMAIL PROTECTED] wrote: I'm trying to connect * server from diax 0.9.8c client and * outputs this errors on CLI Nov 21 18:59:59 NOTICE[7316]: chan_iax2.c:5742 socket_read: Rejected connect attempt from 192.168.0.4, requested/capability 0x2/0x2 incompatible with our capability 0x400. You have a codec problem your * only supports 0x400 (ILBC) the diax requested 0x2 (GSM) so no compatible codec is availiable Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: TDMoE over bonded NIC's
On Sun, 21 Nov 2004 19:50:36 -, Kevin Brennan [EMAIL PROTECTED] wrote: I am planning to configure * box A with PSTN interface to route faxes to * box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for connection between servers. Was wondering - does anybody have experience with TDMoE over bonded interface - ie. does it work ok?. - does anybody have feedback using this scenario for fax? another question, perhaps someone knows what's the limitation for channels on TDMoE interface ? and is there a workaround. I recommend you use Iax trunking rather than TDMoE this would scale better. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] freebsd voicemail everything seems to work??
On Tue, 16 Nov 2004 11:35:32 -, Victor Alvarez [EMAIL PROTECTED] wrote: Hi, Trying to configure a voicemail system on FreeBSD 4.10 + asterisk 0.9.0, I found the following problems: Download the latest asterisk versions from cvs (try a make update in the asterisk src directory) Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can someone tell me what is going on from this debug?
On Fri, 12 Nov 2004 15:50:13 -0500 (EST), Doug Eubanks [EMAIL PROTECTED] wrote: Can someone tell me why Asterisk is sending 404 instead of passing this call to the demo? I have replaced the IPs with descriptions This is the actual asterisk debug, Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'sip.simflex.net' Looking for 19995551212 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found It would appear you do not have 19995551212 as a valid extension in your default context Regards Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Authenticate or DISA?
On Fri, 12 Nov 2004 18:32:55 -0700, Paul Fielding [EMAIL PROTECTED] wrote: I want to authenticate to the phone system, then be able to call an extension or dial an outside line. My preferred method would be to use DISA, because a) it's non-verbal - ie. it doesn't talk, just provides dialtone, and b) it provides dialtone. My alternative seems to be to use Authenticate, and upon authenticating simply send the caller to the appropriate context to punch in extensions or calls. The problem with this is a) it voices the authentication - ie please enter password which to me is inviting people to try to figure it out, and b) after authenticating you don't get a dialtone, just silence. After the Authenticte why not do a Playtones(Dial) this will give dialtone Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple Question
On Sun, 14 Nov 2004 16:44:12 -, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Is this quite simple to set up and can I attach asterix to my landline via a standard modem? Yes no go to http://www.voip-info.org/wiki-Asterisk and read learn try and read try agin Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transferring calls from a Zyxel P2000w
On Mon, 15 Nov 2004 11:25:55 -0500, Chris TenHarmsel [EMAIL PROTECTED] wrote: I don't believe the phone has the ability to transfer calls, I remember looking for this and not finding anything. You need to use # transfer check wiki Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling h@ and Loop Detected
On Fri, 12 Nov 2004 13:39:02 +0100, Nicklas Bondesson [EMAIL PROTECTED] wrote: I see alot of these messages after the line is hung up. Why is that? Urgent handler -- Executing Hangup(SIP/200-9493, ) in new stack Urgent handler -- Executing Dial(SIP/200-9493, SIP/[EMAIL PROTECTED]||T) in new stack Urgent handler Urgent handler -- Called [EMAIL PROTECTED] Urgent handler You seem to have an h conetxt or a wild card in your extensions.conf that is catching the hangup event. You need to post your extensions.conf for us to advise further. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: getting callerid from spa3k to asterisk
You could try adding the line insecure=very to the relevant section of the sip.conf this would force asterisk to only validate the IP address and not the user name (possibly but it is woth a shot) Jason On Mon, 8 Nov 2004 10:28:03 -0800, Randy Bush [EMAIL PROTECTED] wrote: You could maybe look at the autocreatepeer option for sip.conf that level of vulnerability would not seem to be a good approach to solving some sort of sip/config problem :-) the problem is in the sip handshake between the spa3k and *. i have been hoping a sip geek would have a chance to look at it. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Same Extensions in Multiple contexts
On Mon, 8 Nov 2004 20:19:42 -1000, Richard [EMAIL PROTECTED] wrote: I have a question here. If both companies use 200 as their extension, how can * tell which context a received sip call uses? The received sip call will be placed in the context specified buy its definintion in sip.conf Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware Support
Yes look at Ebay for x100P compatible cards On Thu, 04 Nov 2004 10:10:50 -0500, Mike Shultz [EMAIL PROTECTED] wrote: Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dev meeting bridge
Could someone please post the url for the conf? also mute your mic so everyone can hear!!! IAX2/[EMAIL PROTECTED]/4569 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intertex IX66
On Thu, 9 Sep 2004 12:26:56 -0400, Chris HARIGA [EMAIL PROTECTED] wrote: Hi, I have Asterisk w/ 192.168.1.1 and I setup IX66 to be 192.168.1.2 (I'm using pppoe client and dyndns.org on IX66) I setup on Local DNS Server my * box and after that I was able to register my phones from the Internet. I cannot understand my problem with one way sound... what is wrong on my configuration :(( As the IX66 is a sip aware router make sure you have no entries for nat in your sip.conf, and let the ix66 deal with the nat, not * . I hope this helps. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audiocodes Mediant 2000
On Thu, 16 Sep 2004 20:22:57 +0900 (JST), Isamar Maia [EMAIL PROTECTED] wrote: Hi FOlks, I am trying to setup remotely an AudioCodes Mediant 2000 MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Google found this it may help http://corp.deltathree.com/productsandservices/manuals/bizlink.pdf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card
On Thu, 26 Aug 2004 09:25:55 -0600, Andrew Elchuk [EMAIL PROTECTED] wrote: Is a sound card needed in order to playback some of the asterisk sounds in /var/lib/asterisk/sounds when dialing out with an X100P? Thanks. No Sound card is requied ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Channel CLI
On Thu, 26 Aug 2004 17:31:46 +0200, Alessio Focardi [EMAIL PROTECTED] wrote: Also dialing out works like a charm, the only problem is that calling out asterisk is displayed on the called phone instead of the sip address of the asterisk box. In the general section of sip.conf use the following line fromdomain=sip.address.com Regards Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
On Fri, 27 Aug 2004 09:50:37 +0100, Jon Fautley [EMAIL PROTECTED] wrote: Heh, good old BT. I've never tested voice over Business Highway, as every BT engineer/support/sales person I've spoken to swore blind that it wouldn't work - and in BT's eyes, if they say it won't work, it's unsupported, therefore, if it breaks - you're on your own. Also, I don't believe you can get the full range of 'BT Select Services' or whatever they call them today on the Highway lines (things like Call Deflection, and even caller id on the home highway lines, I believe) I use business Highway, (Home highway works but MSN's are not availiable and CLIP- Callerdisplay is not an option for the ISDN Line) I have 8 MSN's Callerdisplay, Plus 2 analogue numbers all works great through a fritz card. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO interfaces used in UK?
On Fri, 27 Aug 2004 11:15:07 +0100, David Gurr [EMAIL PROTECTED] wrote: What FXO interface methods are folks using successfully in the UK? Ditch FXO completely and use a BRI Solution much better quality. or use Digium TDM400P card with two FXO modules, and apply UK CallerID patches, In my opinion ISDN is the way to go. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advice on BT ISDN Services (UK)
On Wed, 25 Aug 2004 14:03:36 +0100, Jon Fautley [EMAIL PROTECTED] wrote: On 25 Aug 2004, at 13:42, Benjamin Johnson wrote: Thanks for that Jon, can anyone confirm whether Asterisk can pick up which MSN has been dialed and route the call depending on this - or does this functionality only work for DDIs. If I have to use DDIs can anyone recommend and active ISDN card which works with Asterisk and is readily available in the UK. I use a BT Speedway card and chan_capi under * with MSN's works fine no issues. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which end hungup?
On Wed, 25 Aug 2004 19:38:34 +0100 (BST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I suspect it's the POTS end since I haven't been able to reproduce it by dialling IAXComm from a SIP client connected to Asterisk 1, but I can't confirm it. What would cause the X100P to randomly drop a call if this is the case? Some sound during the conversation the card has detected as a busy tone set busydetect=no in zapata.conf or increase the busycount=4 to a higer value, if you need busy detection. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems compiling chan_capi-0.3.5
On Tue, 17 Aug 2004 01:16:21 +0200, Patrick [EMAIL PROTECTED] wrote: On Mon, 2004-08-16 at 22:13, Markus Engelbrecht wrote: Hello, so I decided to update to the latest CVS version of asterisk and of chan_capi. You are compiling the wrong version of chan_capi to get chan_capi to work with latest CVS-HEAD you need to uncomment the line in the Makefile as this # if you want to compile against latest (non-stable) asterisk cvs CFLAGS+=-DUNSTABLE_CVS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] First Post: Any existing AVAYA Switch - Asterisk Voicemail configs?
On Thu, 05 Aug 2004 10:09:37 -0400, Mike Cathey
[EMAIL PROTECTED] wrote:
How did you get CID to work? We have a Definity and both an FXO and PRI
(T100P) link to *. We can't seem to get CID to pass at all. We're
running v9{something} on the Definity.
It won't work on the FXO
However it should work fine on the PRI check the definity trunk form
and set send number to y
Jason
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Re: [Asterisk-Users] No incoming audio on incoming SIP calls
On Thu, 5 Aug 2004 16:07:01 +0100, John Howard [EMAIL PROTECTED] wrote:
And my dialstrings look like this:
;Internal lines
exten = 2001,1,Dial(SIP/2001,20,tr)
;outgoing calls
exten = _9XX.,1,Dial(Zap/1/WW${EXTEN:1},60,Tr)
The dial string I am interested in does not seem to be here,
The dial string from the X100p is the one that needs the t
eg
exten = s,1,Dial(SIP/2001SIP/2002,20,tr)
Also ensure in the phone that the DTMF mode is set to
DTMF RELAY inband(RFC2833)
DTMF Payload( 101)
Jason
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Re: [Asterisk-Users] App.c
On Mon, 2 Aug 2004 12:32:38 -0400, AJ Grinnell [EMAIL PROTECTED] wrote: Can someone tell me where I can get just app.c from. Mine somehow got corrupted, and no updates or anything else will fix it. I just need the one file from the latest cvs. 8-1-04. Please help Delete your corrupted app.c and re download from cvs Then make clean make install Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as Front-End for Artisoft Televantage 6
On Mon, 2 Aug 2004 12:54:59 -0700, Alain Bautista [EMAIL PROTECTED] wrote: Anyone had experience 'marrying' the two? We had setup * to front end Artisoft's Televantage. It works with some issues need to be resolved: - Inbound calls could not properly handled and routed by Televantage's Call Classifier. It goes directly to the Televantage's default auto Some more information on how the two systems are connected would help are you using PRI, T1, Analogue etc... Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco PRI no CallerID
On Mon, 02 Aug 2004 20:23:24 -0700, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: * -- SIP -- CISCO -- PRI -- PSTN The PSTN sees no callerid. *--- PRI[zaptel]-- PSTN Callerid is there... which makes me think it's the cisco, not the PRI/PSTN/telco. CISCO PRI-- * PRI [zaptel] Callerid IS there... which makes me shake my head in disbelief, because * can see clid from the cisco pri, but pstn doesn't... but when * sends info on that pri, pstn does see clid. help? It sounds like your Carrier is blocking the CLI on it's PSTN there is nothig you can do about it but talk to them. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] avm c4, ptmp
On Tue, 3 Aug 2004 11:40:28 +0200, Maurizio Marini [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 02 August 2004 18:39, Deti Fliegl wrote: Your Extension has to match your MSNs. You have to configure all MSNs you have in a comma separated list like msn=27849,27852,27869,27861 and you must only use these MSNs as caller id. Hi :) thnx for having tryied to help :) we have 2 number on our isdn: 0721855285 and 0721859609 i try to call my home: 0721950396 here the issue: I would set the MSN's to 855285 and 859609 They do not usually include the area code. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Parking SIP Phones
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Sun, 1 Aug 2004, Trevor Peirce wrote:
Hello,
I know not too long ago I saw /something/ _somewhere_ about an
adjustment to call parking that allowed blind transfers from SIP phones
to park a call and still be able to hear the parking lot stall number.
Unfortunately, I have no idea where I saw that (google turned up little,
couldn't find it on the list either). I'm using Sipura SPA-2000
adapters and it doesn't seem to work with today's CVS.
I use Brian's Valet Parking on our system.
exten = 700,1,ValetParkCall(7${CALLERIDNUM}|mylot|60|${CALLERIDNUM}|1|phones)
exten = _7XX,1,ValetUnParkCall(${EXTEN}|mylot)
To park a call, blind transfer to 700, and to pick it up again, dial
7+your extension. This works well for your small office.
This will only work if you have two digit extensions
Jason
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Re: [Asterisk-Users] Parking SIP Phones
On Mon, 2 Aug 2004 04:50:08 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Mon, 2 Aug 2004, Jason Williams wrote:
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I use Brian's Valet Parking on our system.
exten = 700,1,ValetParkCall(7${CALLERIDNUM}|mylot|60|${CALLERIDNUM}|1|phones)
exten = _7XX,1,ValetUnParkCall(${EXTEN}|mylot)
To park a call, blind transfer to 700, and to pick it up again, dial
7+your extension. This works well for your small office.
This will only work if you have two digit extensions
Jason
exten = 7000,1,ValetParkCall(7${CALLERIDNUM}|mylot|60|${CALLERIDNUM}|1|phones)
exten = _7XXX,1,ValetUnParkCall(${EXTEN}|mylot)
And now it magically works with three digit extensions. Do you need me to
paste the config for four digit extensions as well?
Just try four digit extensions, You will find it is an invalid parking
location. Why don't you try somethig before jumping off the deep end.
Jason
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Re: [Asterisk-Users] Unauthenticated calls from a specific IP
On Fri, 30 Jul 2004 08:56:03 -0400, Deon Rodden [EMAIL PROTECTED] wrote: We put a VWIC and a DSP in a Cisco 1720. The purpose will be for a customer to use a T1 Crossover cable to connect the 1720 into their existing PBX system. It'll be a Virtual T1 PRI type of thing. The Cisco 1720 will make the conversion to SIP and send it to our Asterisk server. As far as his PBX is concerned, it's talking to a standard T1 PRI from the local telco or whatever. The issue is Cisco routers don't support SIP registration/authentication. I want this customer to be in his own context in the extensions.conf file. Add this line to the cisco section insecure=yes ; To match a peer based by IP address only and not peer and make sure the host=xxx.xxx.xxx.xxx is correct Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play CD!
I do that. But when I play MusicOnHold the music is played slowly! I don´t know why... but is how the bitrate is playing with a different number. Make sure you are running mpg123 0.59r and no other version Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT + iConnectHere Broken in 1.0RC1
I've tried setting nat=yes in places, externip, et al with no success .. even though the code I was running from back then worked without that. Some of the options in sip.conf have changed look at the samples in src/asterisk/configs/sip.conf.samples Regards Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call queues
On Fri, 23 Jul 2004 02:26:26 -0400, Jeremy Kenney [EMAIL PROTECTED] wrote: Hello I am new to asterisk I want to setup the call queues where it will ring multiple devices at the same time and send the call to the first one that is picked up. There doesn't need to be an agent login for this I don't think I just want setup so no login is required. Please help Check out the dial command Show application dial dial(device1device2device3) Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 12sp firmware... Anyone got it???
- Original Message - From: Steve McMahon [EMAIL PROTECTED] Date: Fri, 23 Jul 2004 01:12:26 -0700 Subject: [Asterisk-Users] Cisco 12sp firmware... Anyone got it??? To: [EMAIL PROTECTED] Looking for firmware (anything) for the 12sp model phones. Anyone got it drop me a line! www.cisco.com and get a support contract. [EMAIL PROTECTED] or [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Doublehash transfers
On Fri, 23 Jul 2004 12:00:22 -0400, mattf [EMAIL PROTECTED] wrote: That means that you need to hit the pound key twice to initiate a transfer instead of once. Because of our inbound call center we need to do transfers and we also need to be able to hit the pound key once without transferring, so a single pound transfer option is unacceptable. Where did the code go? How can I apply the doublehash patch? I know there are several other people out there that go through what I do every time we res_parking has become res_features so look there somewhere Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-0.3.4b and asterisk last cvs
On Wed, 21 Jul 2004 15:43:17 +0200, Maurizio Marini [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi i've installed asterisk by last cvs and i note res_parking.c is not anymore there; chan_capi-0.3.4b INSTALL file require: in /etc/asterisk/modules.conf insert the line: load = res_parking.so load = chan_capi.so running asterisk i get: [app_capiCD.so]Jul 21 15:32:26 WARNING[1076988448]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber Jul 21 15:32:26 WARNING[1076988448]: loader.c:423 load_modules: Loading module app_capiCD.so failed! how can i fix the issue? 10x for help In /etc/asterisk/modules.conf Insert the line load = res_features.so and remove load = res_parking.so also ensure you have the following in the [global] section [global] chan_modem.so=yes chan_capi.so=yes Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi-0.3.4b and asterisk last cvs
On Thu, 22 Jul 2004 12:10:25 +0200, Diego Ercolani [EMAIL PROTECTED] wrote: Il 10:03, giovedì 22 luglio 2004, Jason Williams ha scritto: ensure you have the following in the [global] section [global] chan_modem.so=yes chan_capi.so=yes sorry, why do you need chan_modem? I don't understand as chan_modem is another channel as are chan_iax, chan_sip Whoopse You only need chan_capi.so=yes I am using chan_modem for other things. Regards Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Daytime - Nighttime
On Thu, 22 Jul 2004 14:54:29 +0100, Steve Hanselman [EMAIL PROTECTED] wrote: Yes, you'd have a dialplan entry that set a value in the database, then acted upon that. You'd probably want some nice voice prompts The system is currently in [Day/Night/Holiday] mode, press 1 to set to day, 2 to set. Here is the start of a simple one I'm sure you will be able to extend it from this http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+autoattendant ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Registration issues
On Tue, 20 Jul 2004 23:50:05 +0200, Andy Powell [EMAIL PROTECTED] wrote: Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I'm using * with an IX66 and no issues, with CVS head I suggest you have a configuration error somewhere it looks like the IX66 is trying to authorise the clients, and no * have you set the IX66 to forward all sip requests for your domain to * ? Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New CVS version
On Tue, 20 Jul 2004 10:49:51 -0400, Seth Remington [EMAIL PROTECTED] wrote: You are probably having a problem with parking being renamed to features. Try a make clean then a make install. If that doesn't work then delete the res_parking.so module from /usr/lib/asterisk/modules/. You may need to change modules.conf to load res_features rather than res_parking ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP client to IAXTel 800/888/877/866 dialing thru Asterisk
On Mon, 12 Jul 2004 11:42:57 -0700, Dameon D. Welch-Abernathy [EMAIL PROTECTED] wrote: Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. Phoneboy IAXcomm use gsm only that may help Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL 2000W
On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager [EMAIL PROTECTED] wrote: Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold, and the right button to take it off - except when I am in a call, these keys have no effect. I've tried teh 000c firmware, the 000e firmware and the Pulver 0011 firmware - but none work, so I'm wondering if this feature just simply isn't implemented, or if there is likely to be something wrong with my asterisk config. No it does not work, you need to use # transfer which will mean you will not be able to dial # into ivr's. Search on wiki for # transfer Regards Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Internal Extenion Config
On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley [EMAIL PROTECTED] wrote:
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
Only a config issue I'm sure
[sip]
exten = 301,1,Dial(SIP/Nick,20,tr)
exten = 302,1,Dial(SIP/Sharon,20,tr)
exten = 1000,1,Dial(SIP/NickSIP/Sharon,20,tr)
exten = 302,2,VoiceMail,u302
exten = 301,2,VoiceMail,u301
exten = 1000,2,VoiceMail,u
exten = 1000,102,VoiceMail,b
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain
include = outgoing
add here
include = internal ; allow sip to dial 310
[incoming]
exten = s,1,Dial(SIP/NickSIP/Sharon,20,tr)
[outgoing]
exten = _7.,1,Dial(IAX2/login:[EMAIL PROTECTED]XXX/${EXTEN:1})
exten = 5.,1,Dial,Zap/1/${EXTEN:1}
[9103]
exten = 21060,1,Dial(SIP/Nick)
exten = 21062,1,Dial(SIP/Sharon)
[internal]
exten = 310,1,Dial,Zap/2
include = sip ; allow internal to dial sip phone
Try those changes and see how you get on
Jason
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Re: [Asterisk-Users] X101P FXO with RED alarm
On Sun, 11 Jul 2004 23:02:56 +0100, Richard Airlie [EMAIL PROTECTED] wrote: On Sat, Jul 10, 2004 at 05:55:21PM +0100, Kevin Walsh wrote: Richard Airlie [EMAIL PROTECTED] wrote: First things first. Scrap the ports and build from the latest CVS source. 0.9 is far to old and buggy, and suspect the same of the Zaptel driver you have, although I don't use *BSD myself. I cvsup'd to the latest source yesterday and tried to build zaptel, but it failed right away. (trying to include linux/*.h) You need to get zaptel built correctly with your kernel otherwise it will never run correctly. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 configuration, getting help via IRC?
On Tue, 6 Jul 2004 11:58:58 -0500, Paul Concepcion [EMAIL PROTECTED] wrote: Loopback should always make your status LEDs glow steady green. If that's not working then you've got other problems. It seems I may have those other problems you talked about. I made a loopback cable and tested it on the channel bank. After about three seconds all the status lights went green. I plugged it into the T100P with varying effects. I was grasping a little, and tried different first lines of the /etc/zaptel.conf file: span=1,0,0,esf,b8zs OR span=1,1,0,esf,b8zs cycles between: RED- YEL/RED - YEL/REC - Red/REC - OK. Eventually settles into RED. Looks like you have a card problem a loop back to yhe T100P should go green in about 3 seconds like the channel bank. Regards Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1102 Problems
On Tue, 6 Jul 2004 13:37:33 -0500, McInnis, JP [EMAIL PROTECTED] wrote: We have a Mediatrix 1102 hooked into the network. Both of the attached analog phones and all of their features work, but in the CLI we keep getting -- Got SIP response 481 Transaction Does Not Exist back from XXX.XXX.XXX.XXX (Where XXX is the IP address of the Mediatrix ) every few minutes. I have changed most of the settings in the sip.conf multiple times and have done multiple cvs updates. Any help with this would be most appreciated. [602] type=friend secret=blah username=602 context=default host=dynamic canreinvite= no qualify=200 dtmfmode=inband defaultip=XXX.XXX.XXX.XXX callerid=MediaTrix Port 1 602 mailbox=602 I would remove the mailbox line as the voicemail notifications may well be causing the problem Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
