Re: [Asterisk-Users] record a conversation
show application monitor in the cli Matteo. Il mer, 2003-07-02 alle 09:22, Herv Thibaud ha scritto: hi is there a simple way to record a conversation with asterisk ? thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Enhanced queue app
Benjamin Miller wrote: Would it be more flexible to approach this differently, with a dtmf to indicate that the agent is done with wrap up? So they get off a call and can wrap up the call for as long as necessary, and then hit * or something that marks them as available again rather than working against a timer to get a call wrapped up in 30 seconds or something, or shorting the timer because its always to long or something? Just my $.02 on the topic. Ben I`m looking forward to using * as a call center solution in future, and really excited about this thing. BUt I`m ashamed to say tham I`m not familuar with the terms, and what do wrap-up mean? -Original Message- From: Jim Friedeck [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 3:52 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Enhanced queue app Mark, How hard would it be to write a simple app to cancel wrap-up via an extension? Like dialing *99 to short- out the remaining wrap-up time? Jim Friedeck --- Mark Spencer wrote: Should wrap up time be something associated with a queue, or with an agent? Mark On Tue, 1 Jul 2003, Jim Friedeck wrote: Will try to change to this: Agent picks up phone and dials extension to 'login app': exten = 750,1,QueueLogin(QueueName, wrap-up-time) This would allow for quick agents to log into a queue for faster processing and allow slower processing for slow agents. An agent would simply log out if more time was needed. I could not think of a quick way to cancel wrap-up waiting. Our Inter-Tel has a programmable wrap-up cancel button. I don't think this would be very easy on POTS phones. Any ideas? Jim Friedeck -- TC wrote: I have also contracted mark for some minor modifications to app_queue and chan_agent 1) if you use a mixed environment of agents and devices on a single Q I want the ring process terminated before the time out value in queue.conf if the call is picked up by anyone assigned to the Q (device or agent) 2) if all agents are online when a new call comes into a Q, the current q logic will ring the devices for the timeout period before cycling and attempting to assign the called to a logged in agent , I want the Q to attempt to assign a call to an agent as soon as they hit the * key to hang up on the current call even if the ring process has started on the devices I also have some in line comments here see %TC To all who need more queue functionality, We are contracting Digium to enhance the queue app for our call center needs. Please read the following email conversation and give your ideas. Unless a glaring omission is found in my specification we will have them start tomorrow (Wednesday). I may not have thought of something important. It will be released to all Asterisk users by Digium. Thanks for your time. %TC THANK YOU JIM If agent recieves call while logged in and call goes unanswered for a specified amount of times (specified per queue) agent is logged out and event is recorded in CDR. Notification through astman interface would be desireable as well for management purposes. %TC Can we just make sure that specified amount of times has a value 0 zero meaning forevever to stop agents automagically beling logged out by the system When agent picks up phone and is not on a local interface, a per-queue option to ask for confirmation by pressing a DTMF digit. This tells queue that call will be handled by this agent. %TC Can this option be a configuration of the agent.conf NOT queue.conf seems to me that it is the agent who would like that discretion not the queue process that should enforce this rule (I have already hacked this feature in chan_agent) If not confirmed, a per-queue option to log agent out or skip and place agent at bottom of queue. (Not really necessary but I could see it being useful for agents working from home with kids.) %TC again should this not be an agent.conf issue. also can we make sure the flag allows for 1-skip don't change agent place in Q 2-skip force to bottom of the Q 3-log them out Calls would be routed to the agent who took a call successfully longest ago. This would be the fairest way to distribute them to the busiest people. People on a call unrelated to that queue would maintain their position in the queue order unless they logged out. A busy agent could be making outbound calls and it would be unfair to penalize them for being unavailable due to outbound activity. Perhaps a per-queue choice for this. A per-queue specified delay after hanging up that would
Re: [Asterisk-Users] record a conversation
and what the way to play records in the spool Le mer 02/07/2003 à 09:28, Matteo Brancaleoni a écrit : show application monitor in the cli Matteo. Il mer, 2003-07-02 alle 09:22, Hervé Thibaud ha scritto: hi is there a simple way to record a conversation with asterisk ? thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with echo
I had a similar problem and solved it changing the params of input gain on my pstn-gateway, change from a value of 10 to a value of 1 and that eliminated the echo on the SIP Phones. Dave Packham wrote: Same prob here. 15 SIP phones only get eco when going to the PSTN... if you find something let me know Dave [EMAIL PROTECTED] 7/1/2003 8:53:13 AM Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A solution for SIP and NAT
That's a different part of the equation. If Asterisk could interpret the Via: headers like the Cisco phones do, that would solve the Asterisk-behind-a-NAT problem to a large degree. Perhaps it already does; I've never tried putting Asterisk behind a NAT, only SIP clients. JT Please don't take the discussion of SIP interactions off list. I already have NATed SIP clients working with *, but * still has problems where its own external IP is not public and it is trying to use external SIP services. A full discussion on list could spawn an Asterisk SIP FAQ - and I think that would be a good thing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A solution for SIP and NAT
On Wed, 2003-07-02 at 04:26, John Todd wrote: You may be correct about the Via: header, but you're incorrect in the concept as to how it relates to Asterisk, notably in your reversal of what side of the transaction is putting data in the Via: header to make SIP work correctly. This is cluttering up the list. Talk to me off line if you want a better understanding of how NAT and SIP work with Cisco devices. Again, for those of you who might be trying to figure out what the result of this conversation is: SIP clients behind NAT works fine in both directions (incoming and outgoing calls), Asterisk makes it work, it's not using STUN. Cisco devices work especially well. JT Hi John, Thanks for the very helpful info so far. I concur with Richard Alexander's request to keep this discussion on list. How about Asterisk and NAT? Can you please comment if the examples below also work. 1x SIP phone - NAT box - Internet - NAT box - Asterisk 10x SIP phone - NAT box - Internet - NAT box - Asterisk The SIP phone(s) and Asterisk server are on private IP addresses. The NAT boxes (e.g. adsl router) have a public IP address. Any requirements for the NAT boxes like being a SIP proxy? Thanks, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A solution for SIP and NAT
Date: Tue, 1 Jul 2003 14:37:20 +1000 From: Andrew Radke [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] A solution for SIP and NAT ... So I've started a really simple SIP and RTP proxy project, SaRP, on sourceforge.net. Yesterday we uploaded 0.2 of the perl based release. This is the first general release and should work for most people. We are using it quite successfully for standard calls between all sorts of NATed clients. All you need to do is forward UDP/5060 from your firewall/router to the box running SaRP if you want incoming calls to work and also allow UDP traffic from the ports listed in the config file out. The project can be found at http://sarp.sourceforge.net/ There is also a similar project called siproxd: http://sourceforge.net/projects/siproxd/ regards, klaus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with echo
Hi, What do you mean by pstn-gateway? There is no input gain parameter in zapata.conf file? It is about rxgain? BR, Dan - Original Message - From: Ing. Angel Gomez Garcia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:48 AM Subject: Re: [Asterisk-Users] Problem with echo I had a similar problem and solved it changing the params of input gain on my pstn-gateway, change from a value of 10 to a value of 1 and that eliminated the echo on the SIP Phones. Dave Packham wrote: Same prob here. 15 SIP phones only get eco when going to the PSTN... if you find something let me know Dave [EMAIL PROTECTED] 7/1/2003 8:53:13 AM Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A solution for SIP and NAT
Patrick wrote: [snip] Hi John, [snip] How about Asterisk and NAT? Can you please comment if the examples below also work. 1x SIP phone - NAT box - Internet - NAT box - Asterisk 10x SIP phone - NAT box - Internet - NAT box - Asterisk This all depends on the NAT boxes that you use. The SIP phone. Whether the call is going out from or into the SIP phone. Whether the SIP phone is registered with Asterisk ot just making a call to it. Etc. The SIP phone(s) and Asterisk server are on private IP addresses. The NAT boxes (e.g. adsl router) have a public IP address. Any requirements for the NAT boxes like being a SIP proxy? Some brands of router will handle this virtually transparently especially if you register with the Asterisk calls and all conversations are directly with it. Thanks, Patrick Regards, Andrew Radke. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Hot Desks??
Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after successfully validating the user is then prompted for phone number (being some IP phone ID number or an external Mobile or Home phone number).. All calls made to that users DID number or extension are now routed to the registered destination device.. Any calls the user makes from any IP Phones carry the correct caller ID information as well.. Anyone got something like this or any form of user manageable extension control or hot desk type solution working?? Or any suggestions how it could be archived?? Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with echo
I have a SIP FXO 8 port VoIP gateway, and it has a parameter called 'input gain' wich is the one I modified, there might be a similar parameter on the configuration for the hardware you are using. Dan wrote: Hi, What do you mean by pstn-gateway? There is no input gain parameter in zapata.conf file? It is about rxgain? BR, Dan - Original Message - From: Ing. Angel Gomez Garcia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:48 AM Subject: Re: [Asterisk-Users] Problem with echo I had a similar problem and solved it changing the params of input gain on my pstn-gateway, change from a value of 10 to a value of 1 and that eliminated the echo on the SIP Phones. Dave Packham wrote: Same prob here. 15 SIP phones only get eco when going to the PSTN... if you find something let me know Dave [EMAIL PROTECTED] 7/1/2003 8:53:13 AM Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A solution for SIP and NAT
Klaus Darilion wrote: [snip] The project can be found at http://sarp.sourceforge.net/ There is also a similar project called siproxd: http://sourceforge.net/projects/siproxd/ regards, klaus It has a broadly similar goal on the surface but a very very different approach. siproxd relies on a large library. It tries to be completely SIP compliant (you can't be compliant and handle NAT). It leaves outgoing RTP traffic to go direct with inbound coming to it. It requires ports to be forwarded to it/opened for incoming RTP traffic (SaRP doesn't). It doesn't consider security. And lastly myself and a number of other people just haven't seen it work. There are a couple of us that now actually use SaRP to develop SaRP (the old the compiler can compile itself test :-). It handles any sort of NAT that can get the SIP port to it (5060 or whatever else you want). It drops any packets that don't make sense and logs them; don't reply unless you understand a packet or you are just giving away information about your network or opening yourself up to an attack. And it's cross-platform. There is no way you'll get siproxd to run on Windows for example. I'm not taking anything away from siproxd, I'm just stating why I don't use it and why I don't know anyone who does. Regards, Andrew Radke ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with musiconhold
Hi evereybody,
I'm trying to use musiconhold during dial
tones.But I only can call earing dial tones instead of music.
Now will see my configuration files.
AGI File(using AGI script to EXEC
DIAL)
print "EXEC Dial Zap/g2/numberc||m\";
$res=checkresult();
Extension.conf
exten =_numberb,1,Answerexten
=_numberb,2,SetMusicOnHold,defaultexten
=_numberb,3,AGI,dial.agi
MusicOnHold.conf
default =
quietmp3:/var/lib/asterisk/mohmp3,-z
Note: Inside the directory
/var/lib/asterisk/mohmp3/ there are severals mp3 files.
Log File
(/var/log/asterisk/debug)
Jul 2 12:27:34 DEBUG[630802]: File
app_dial.c, Line 333 (dial_exec): SIMPLE DIAL (NO URL)Jul 2 12:27:39
DEBUG[630802]: File chan_zap.c, Line 1936 (zt_setoption): Set option AUDIO MODE,
value: ON(1) on Zap/2-1Jul 2 12:27:39 DEBUG[630802]: File chan_zap.c,
Line 1493 (zt_hangup): Hangup: channel: 2 index = 0, normal = 13, callwait = -1,
thirdcall = -1Jul 2 12:27:39 DEBUG[630802]: File chan_zap.c, Line 1846
(zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/2-1Jul 2
12:27:39 DEBUG[630802]: File chan_zap.c, Line 908 (update_conf): Updated
conferencing on 2, with 0 conference usersJul 2 12:27:39
DEBUG[630802]: File chan_zap.c, Line 1930 (zt_setoption): Set option AUDIO MODE,
value: OFF(0) on Zap/2-1Jul 2 12:27:39 DEBUG[630802]: File app_agi.c,
Line 1216 (run_agi): Zap/1-1 hungupJul 2 12:27:39 DEBUG[630802]: File
cdr_mysql.c, Line 58 (mysql_log): cdr_mysql: inserting a CDR
record.Jul 2 12:27:39 DEBUG[630802]: File cdr_mysql.c, Line 61
(mysql_log): cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2003-07-02 12:27:39','numbera','numbera','numberb','default',
'Zap/1-1','','Dial','Zap/g2/numberc||m',5,5,4,3,'')Jul 2 12:27:39
DEBUG[630802]: File chan_zap.c, Line 1936 (zt_setoption): Set option AUDIO MODE,
value: ON(1) on Zap/1-1Jul 2 12:27:39 DEBUG[630802]: File chan_zap.c,
Line 1493 (zt_hangup): Hangup: channel: 1 index = 0, normal = 12, callwait = -1,
thirdcall = -1Jul 2 12:27:39 DEBUG[630802]: File chan_zap.c, Line 1846
(zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/1-1Jul 2
12:27:39 DEBUG[630802]: File chan_zap.c, Line 908 (update_conf): Updated
conferencing on 1, with 0 conference usersJul 2 12:27:39
DEBUG[630802]: File chan_zap.c, Line 1930 (zt_setoption): Set option AUDIO MODE,
value: OFF(0) on Zap/1-1
If somebody can help I will be very pleasured, now
I'm really lost.
Thks a lot.
Re: [Asterisk-Users] Asterisk and Hot Desks??
Actually its easier than you think... Allocate a control extension for each hot desk user. implement call forward and cancel call forward... I did this with a ifo storage in the astdb so it holds during a restart... your dial plan must be macro'd for this to work properly... have a look at junghans site (I will get around to publishing my complete solution when I finally sort all the functions out On Wed, 02 Jul 2003 10:21:47 +, WipeOut . wrote: Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after successfully validating the user is then prompted for phone number (being some IP phone ID number or an external Mobile or Home phone number).. All calls made to that users DID number or extension are now routed to the registered destination device.. Any calls the user makes from any IP Phones carry the correct caller ID information as well.. Anyone got something like this or any form of user manageable extension control or hot desk type solution working?? Or any suggestions how it could be archived?? Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Hot Desks??
How about the logon wizard of the snom 100? I think that does something simlar to what you want. It's designed to allow different people to login to a single phone. - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:21 AM Subject: [Asterisk-Users] Asterisk and Hot Desks?? Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after successfully validating the user is then prompted for phone number (being some IP phone ID number or an external Mobile or Home phone number).. All calls made to that users DID number or extension are now routed to the registered destination device.. Any calls the user makes from any IP Phones carry the correct caller ID information as well.. Anyone got something like this or any form of user manageable extension control or hot desk type solution working?? Or any suggestions how it could be archived?? Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Hot Desks??
The snom phones (and I assume others) allow you to have multiple SIP accounts on a single phone. The user logs in to the phone which logs in *. The downside is that you can only log in to accounts set up on the phone rather than any account set up on * but is useful for shared desks etc.. W - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:21 AM Subject: [Asterisk-Users] Asterisk and Hot Desks?? Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after successfully validating the user is then prompted for phone number (being some IP phone ID number or an external Mobile or Home phone number).. All calls made to that users DID number or extension are now routed to the registered destination device.. Any calls the user makes from any IP Phones carry the correct caller ID information as well.. Anyone got something like this or any form of user manageable extension control or hot desk type solution working?? Or any suggestions how it could be archived?? Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with echo
rxgain and txgain are used, for example with the X100P. As I understand it, the echo problem with a SIP to PSTN implementation in * has two components: - echo resulting from the digital to analogue conversion at the X100P - acoustic feedback within the handset used The former is reduced by using the zaptel echo canceller set by this in zapata.conf: echocancel=yes echocancelwhenbridged=yes The choice of echo canceller to use is made when you compile zaptel. mec2 is the default. You can enable aggressive cancellation in mec2 but this can be a bit too severe making calls sound almost half duplex. Mec3 seems to be a bit unstable. You can reduce feedback related echo by tuning rxgain and/or txgain. A value of -3.0 will set the gain at about 70% of its initial value. Iain --On Wednesday, July 2, 2003 3:40 am -0700 Ing. Angel Gomez Garcia [EMAIL PROTECTED] wrote: I have a SIP FXO 8 port VoIP gateway, and it has a parameter called 'input gain' wich is the one I modified, there might be a similar parameter on the configuration for the hardware you are using. Dan wrote: Hi, What do you mean by pstn-gateway? There is no input gain parameter in zapata.conf file? It is about rxgain? BR, Dan - Original Message - From: Ing. Angel Gomez Garcia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:48 AM Subject: Re: [Asterisk-Users] Problem with echo I had a similar problem and solved it changing the params of input gain on my pstn-gateway, change from a value of 10 to a value of 1 and that eliminated the echo on the SIP Phones. Dave Packham wrote: Same prob here. 15 SIP phones only get eco when going to the PSTN... if you find something let me know Dave [EMAIL PROTECTED] 7/1/2003 8:53:13 AM Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX Billing
Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with echo
Hi, In zapata.conf I have tried to change the rxgain and txgain parameters, but without any success. I think it is X100P card driver related issue. BR, Dan - Original Message - From: Ing. Angel Gomez Garcia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 1:40 PM Subject: Re: [Asterisk-Users] Problem with echo I have a SIP FXO 8 port VoIP gateway, and it has a parameter called 'input gain' wich is the one I modified, there might be a similar parameter on the configuration for the hardware you are using. Dan wrote: Hi, What do you mean by pstn-gateway? There is no input gain parameter in zapata.conf file? It is about rxgain? BR, Dan - Original Message - From: Ing. Angel Gomez Garcia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:48 AM Subject: Re: [Asterisk-Users] Problem with echo I had a similar problem and solved it changing the params of input gain on my pstn-gateway, change from a value of 10 to a value of 1 and that eliminated the echo on the SIP Phones. Dave Packham wrote: Same prob here. 15 SIP phones only get eco when going to the PSTN... if you find something let me know Dave [EMAIL PROTECTED] 7/1/2003 8:53:13 AM Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Billing
ok, I'll bite :-) What the heck is a phone shop system ?? On Wed, 02 Jul 2003 09:48:44 +, shepherd fungayi wrote: Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference calls
You may want to check, but I think rtc is a x86ism and may not be available to you on a mac. On Wed, 2003-07-02 at 00:54, Serge Mankovski wrote: I am trying to compile it under Yellow Dog 3.0 on iMac I get this error zaprtc.c:1077: warning: implicit declaration of function `barrier' zaprtc.c:1078: warning: implicit declaration of function `cpu_relax' zaprtc.c: At top level: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Any idea why? Thanks, Serge From: Klaus-Peter Junghanns [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Conference calls Date: 01 Jul 2003 15:38:37 +0200 Hi, if you dont have usb-uhci you can also use your realtime clock to generate zaptel timing. Make sure you dont have rtc support compiled into your kernel and grab zaprtc from: http://www.junghanns.net/asterisk regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Die, 2003-07-01 um 16.29 schrieb Martin Pycko: You need to look at show application meetme in the asterisk CLI but for it to work you need to have some kind of zaptel hardware or emulate it with zttdummy (but for that you need to have usb-uhci like USB controller) and then exten = 1000,1,Meetme,1000 Martin On Tue, 1 Jul 2003, Serge Mankovski wrote: Hi I want to set up * as a conference bridge. I would like to be able to conference is SIP calls (up to 12) I am looking through all available documentation for * to get info on how it is done. No luck so far. Can somebody direct me to the info in this subject? Thank you Serge _ Protect your PC - get McAfee.com VirusScan Online http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Hot Desks??
The problem with this solution is that you are moving the logic to the phone which can very quickly become an admin headache.. the more the config and admin can be central and server based the better.. How about the logon wizard of the snom 100? I think that does something simlar to what you want. It's designed to allow different people to login to a single phone. - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:21 AM Subject: [Asterisk-Users] Asterisk and Hot Desks?? Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after successfully validating the user is then prompted for phone number (being some IP phone ID number or an external Mobile or Home phone number).. All calls made to that users DID number or extension are now routed to the registered destination device.. Any calls the user makes from any IP Phones carry the correct caller ID information as well.. Anyone got something like this or any form of user manageable extension control or hot desk type solution working?? Or any suggestions how it could be archived?? Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soft SIP phones (with RING !!)
Hi all, I have the M1500 Plantronics now and have done some tests with the Mitsumi BT USB adapter. The latest drivers from the Mitsumi site supports Headset Profile too, but I still cannot use it with my dongle (cannot be activated). Anyone else succeeded in using a BT dongle with Headset Profile supports? The included BT receiver works very well, but it would be very nice to use it directly with a BT dongle, (not using the sound card) with SJPhone or X-Lite. Thanks, Dan - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 17, 2003 5:32 PM Subject: Re: [Asterisk-Users] Soft SIP phones (with RING !!) Hi Gary, Mmm, but then I also have to have a sound card in the PC !! It seems that some new drivers for the BT-USB dongle supports Headset Profile too. I still wait for my M-1500 and then I will make a test with my Mitsumi (TDK drivers) BT-USB dongle. I'll keep you in touch if you are interested. Best regards, Dan - Original Message - From: Gary [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 17, 2003 1:35 PM Subject: Re: [Asterisk-Users] Soft SIP phones (with RING !!) On Sat, 17 May 2003 08:34:47 +0300, Dan wrote: Hi Gary, I am also looking for a USBBluetooth adapter which will work well with a a bluetooth headset with my mobile phone. Have you tried Plantronics M-1500? It is a Bluetooth headset with a bluetooth receiver with a 2.5mm jack for both mic and headphone. It can be used as BT headset with your PC through the soundcard. BR, Dan Mmm, but then I also have to have a sound card in the PC !! Considering that the machine I have this RDP session initiated from doesn't have ANY decent sound drivers which actually work with XP a USB headset is wanted, now if i could combine blue tooth and usb for sound from the computer that would be great, but alas, its seems not to be ;-( Gary . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Seg Fault!!
Hi, I have just updated to the latest CVS version ( Approx 13:05 GMT Today ).. I am running chan_capi 0.2.2.. When a call is received Asterisk seg faults.. Not sure what information would be usefull to post so let me know what info will help to debug the problem.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX Billing
I'm very interested in the same thing for a hotel system I would like to implement. Anyone know if the country codes be tied to a pricing lookup table? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 5:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seg Fault!!
Nevermind... After a reboot it appears to be happy again.. must just be a gremlin that crept in somewhere.. Later.. Hi, I have just updated to the latest CVS version ( Approx 13:05 GMT Today ).. I am running chan_capi 0.2.2.. When a call is received Asterisk seg faults.. Not sure what information would be usefull to post so let me know what info will help to debug the problem.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference calls
Never mind my previous post, A quick gogle search shows there is rtc on ppc arch. On Wed, 2003-07-02 at 07:08, Ing. Angel Gomez Garcia wrote: Hi. How can I know if rtc support is built into the kernel ? Steven Critchfield wrote: You may want to check, but I think rtc is a x86ism and may not be available to you on a mac. On Wed, 2003-07-02 at 00:54, Serge Mankovski wrote: I am trying to compile it under Yellow Dog 3.0 on iMac I get this error zaprtc.c:1077: warning: implicit declaration of function `barrier' zaprtc.c:1078: warning: implicit declaration of function `cpu_relax' zaprtc.c: At top level: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Any idea why? Thanks, Serge From: Klaus-Peter Junghanns [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Conference calls Date: 01 Jul 2003 15:38:37 +0200 Hi, if you dont have usb-uhci you can also use your realtime clock to generate zaptel timing. Make sure you dont have rtc support compiled into your kernel and grab zaprtc from: http://www.junghanns.net/asterisk regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705392 fax: +49 30 79705391 iaxtel:1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Die, 2003-07-01 um 16.29 schrieb Martin Pycko: You need to look at show application meetme in the asterisk CLI but for it to work you need to have some kind of zaptel hardware or emulate it with zttdummy (but for that you need to have usb-uhci like USB controller) and then exten = 1000,1,Meetme,1000 Martin On Tue, 1 Jul 2003, Serge Mankovski wrote: Hi I want to set up * as a conference bridge. I would like to be able to conference is SIP calls (up to 12) I am looking through all available documentation for * to get info on how it is done. No luck so far. Can somebody direct me to the info in this subject? Thank you Serge _ Protect your PC - get McAfee.com VirusScan Online http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A solution for SIP and NAT
Instead of this make notes of some of the faults in SIP that cause you problems and start working towards SIP/2.1 or SIP/3.0. Just because you weren't one of the people involved in designing the existing protocol doesn't mean you can't work to change it. SIP 2.0 has some unbeleivably braindead concepts in it. It is so loose that you can find one peice of info in half a dozen places in a SIP packet. It has no tightly defined structure and has no concept of how to work in a real-world network. Security wasn't truly even an afterthought which in the modern Internet environment is disgraceful and then there are the reasons you've given below. This should not mean we just kludge everything together. A lot of stuff can be tidied up significantly and at least some of it can be thrown out. As such we should be working towards getting a new draft out that doesn't mean throwing out existing infrastructure but does allow for SIP/VoIP to move forward on the Internet not just corporate intranets. Of course getting IAX accepted as an Internet draft and moving everyone on it would probably be easier than fixing SIP :-) but you fight the (small) battles you can win. Sorry if this sounds like a rant. Regards, Andrew Radke Michael Kane wrote: At the end of the day we all probably can get SIP and NAT to work together if we spend TIME configuring our NAT boxes and SIP devices to negotiate the traversal of a NAT. In the end result, the WAN IP must be is correctly added to the contact table(sipd) or location table(SER), allowing the proxy to route a call destined for that UA. Now, my delima as a service provider, is how do I document this for every SIP device out there where my mother can purchase a UA device, plug it in, and start placing calls without putting on a poodle suit and jump through flaming hoops. That's why(for me) it becomes an operational nightmare, not only to document vendor configs(if they support NAT traversal), but, then support the end user on how to config their devices. That why I have looked into(implemented) such technologies like STUN and probably will be forced to purchase a SIP aware firewall that will spoof and re-arrange SIP messages destined for my proxy server. Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A solution for SIP and NAT
Andrew Radke wrote: Ok I guess it's time for me to weigh in on this since I started the whole thing and am the main developer of SaRP. NAT and SIP _can_ work okay under very very restricted circumstance. Multiple SIP UAs behind one NATed IP _can_ work okay with a very intelligent router/firewall. [deleted] When Asterisk is running on the same device as NAT, does this make the device an intelligent router/firewall? In other words, is the SIP proxy in Asterisk smart enough to translate IP address tcp/udp ports as needed? Will it also add/remove rules in the fw (or provide a hook for an external function/script to do so?) Thanks, MikeC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX Billing
Shepherd- Having designed one of these in the past (in a higher level voice environment), I can tell you that this is not a small undertaking. It's at least as much an SQL job as a voice task. Usually the way to accomplish this is to establish more-or-less a pre-paid phone card system, where the shop prepays an overall amount for international calling access. Then you have to time each call as it is occurring, debiting each account, and the master account, in real-time. This can be a bit complex when you have 20 or 30 calls going at one time. You have to cut them off promptly when the money runs out (big problem). And you have to provide call detail and charges to them at the end of each call, using their own retail tariff. To add to the complexity, each country has a different tariff from the long distance carrier, and within the country, major cities often have special rates per minute. Mobiles have a different tariff too. Phone card platforms usually include a least-cost routing system which chooses a carrier real time based on the call. Tariffs change weekly and must be updated in the system. Anyway, I'm just scratching the surface! I'll write more when I can! Cheers Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Enhanced queue app
That would be excellent. Thanks. Jim Friedeck Mark Spencer wrote: Could probably make '#' terminate wrapup time immediately or something. Mark On Tue, 1 Jul 2003, Jim Friedeck wrote: Mark, How hard would it be to write a simple app to cancel wrap-up via an extension? Like dialing *99 to short- out the remaining wrap-up time? Jim Friedeck --- Mark Spencer wrote: Should wrap up time be something associated with a queue, or with an agent? Mark On Tue, 1 Jul 2003, Jim Friedeck wrote: Will try to change to this: Agent picks up phone and dials extension to 'login app': exten = 750,1,QueueLogin(QueueName, wrap-up-time) This would allow for quick agents to log into a queue for faster processing and allow slower processing for slow agents. An agent would simply log out if more time was needed. I could not think of a quick way to cancel wrap-up waiting. Our Inter-Tel has a programmable wrap-up cancel button. I don't think this would be very easy on POTS phones. Any ideas? Jim Friedeck -- TC wrote: I have also contracted mark for some minor modifications to app_queue and chan_agent 1) if you use a mixed environment of agents and devices on a single Q I want the ring process terminated before the time out value in queue.conf if the call is picked up by anyone assigned to the Q (device or agent) 2) if all agents are online when a new call comes into a Q, the current q logic will ring the devices for the timeout period before cycling and attempting to assign the called to a logged in agent , I want the Q to attempt to assign a call to an agent as soon as they hit the * key to hang up on the current call even if the ring process has started on the devices I also have some in line comments here see %TC To all who need more queue functionality, We are contracting Digium to enhance the queue app for our call center needs. Please read the following email conversation and give your ideas. Unless a glaring omission is found in my specification we will have them start tomorrow (Wednesday). I may not have thought of something important. It will be released to all Asterisk users by Digium. Thanks for your time. %TC THANK YOU JIM If agent recieves call while logged in and call goes unanswered for a specified amount of times (specified per queue) agent is logged out and event is recorded in CDR. Notification through astman interface would be desireable as well for management purposes. %TC Can we just make sure that specified amount of times has a value 0 zero meaning forevever to stop agents automagically beling logged out by the system When agent picks up phone and is not on a local interface, a per-queue option to ask for confirmation by pressing a DTMF digit. This tells queue that call will be handled by this agent. %TC Can this option be a configuration of the agent.conf NOT queue.conf seems to me that it is the agent who would like that discretion not the queue process that should enforce this rule (I have already hacked this feature in chan_agent) If not confirmed, a per-queue option to log agent out or skip and place agent at bottom of queue. (Not really necessary but I could see it being useful for agents working from home with kids.) %TC again should this not be an agent.conf issue. also can we make sure the flag allows for 1-skip don't change agent place in Q 2-skip force to bottom of the Q 3-log them out Calls would be routed to the agent who took a call successfully longest ago. This would be the fairest way to distribute them to the busiest people. People on a call unrelated to that queue would maintain their position in the queue order unless they logged out. A busy agent could be making outbound calls and it would be unfair to penalize them for being unavailable due to outbound activity. Perhaps a per-queue choice for this. A per-queue specified delay after hanging up that would allow agent to get ready for the next incoming call. This might be deactivated by agent dialing 'ready app' or some other convenient way. %TC again is this not realy a configuration item for the agent not the queue process ??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] H.323 Gateway Connection
Hi Justin, Try: exten=242,1,Dial(h323/[EMAIL PROTECTED]) Regards, Szymon Czyz Justin Eckhouse [EMAIL PROTECTED] wrote: Hi, I'm trying to setup Asterisk to allow users to dial out to the PSTN using a remote box supporting h.323. I'm using chan_h323.so, and I'm able to make outbound calls to a client like netmeeting with a line like this: exten = 242,1,Dial(h323/xxx.xxx.xxx.xxx) And I'm able to receive incoming calls to asterisk. However I'm not sure how to route calls to the remote h.323 gateway. In my nave state I've tried something like this (xxx is the IP of the h.323 gw): exten = 244,1,Dial(h323/xxx.xxx.xxx.xxx/PSTN-NUMBER-HERE) When I dial 244, nothing happens, this appears in the console: -- Called xxx.xxx.xxx.xxx == No one is available to answer at this time Any pointers in the right direction would be greatly appreciated. Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Enhanced queue app
Wrap-up, as our existing phone system calls it, is a period of time that an agent will not get an incoming call after hanging up the previous call. This allows time for the agent to 'wrap-up' the preceeding conversation by filling out forms, typing on the computer, or taking a sip of coffee. Most systems have a variable wrap-up time for different agents or queues and a way for the agent to indicate he/she is immediately ready to take a call (like a programmable button on a digital phone.) Since Asterisk deals with analog phones, there is no programmable button we can use. Jim Friedeck -- Anton Yurchenko wrote: Benjamin Miller wrote: Would it be more flexible to approach this differently, with a dtmf to indicate that the agent is done with wrap up? So they get off a call and can wrap up the call for as long as necessary, and then hit * or something that marks them as available again rather than working against a timer to get a call wrapped up in 30 seconds or something, or shorting the timer because its always to long or something? Just my $.02 on the topic. Ben I`m looking forward to using * as a call center solution in future, and really excited about this thing. BUt I`m ashamed to say tham I`m not familuar with the terms, and what do wrap-up mean? -Original Message- From: Jim Friedeck [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 3:52 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Enhanced queue app Mark, How hard would it be to write a simple app to cancel wrap-up via an extension? Like dialing *99 to short- out the remaining wrap-up time? Jim Friedeck --- Mark Spencer wrote: Should wrap up time be something associated with a queue, or with an agent? Mark On Tue, 1 Jul 2003, Jim Friedeck wrote: Will try to change to this: Agent picks up phone and dials extension to 'login app': exten = 750,1,QueueLogin(QueueName, wrap-up-time) This would allow for quick agents to log into a queue for faster processing and allow slower processing for slow agents. An agent would simply log out if more time was needed. I could not think of a quick way to cancel wrap-up waiting. Our Inter-Tel has a programmable wrap-up cancel button. I don't think this would be very easy on POTS phones. Any ideas? Jim Friedeck -- TC wrote: I have also contracted mark for some minor modifications to app_queue and chan_agent 1) if you use a mixed environment of agents and devices on a single Q I want the ring process terminated before the time out value in queue.conf if the call is picked up by anyone assigned to the Q (device or agent) 2) if all agents are online when a new call comes into a Q, the current q logic will ring the devices for the timeout period before cycling and attempting to assign the called to a logged in agent , I want the Q to attempt to assign a call to an agent as soon as they hit the * key to hang up on the current call even if the ring process has started on the devices I also have some in line comments here see %TC To all who need more queue functionality, We are contracting Digium to enhance the queue app for our call center needs. Please read the following email conversation and give your ideas. Unless a glaring omission is found in my specification we will have them start tomorrow (Wednesday). I may not have thought of something important. It will be released to all Asterisk users by Digium. Thanks for your time. %TC THANK YOU JIM If agent recieves call while logged in and call goes unanswered for a specified amount of times (specified per queue) agent is logged out and event is recorded in CDR. Notification through astman interface would be desireable as well for management purposes. %TC Can we just make sure that specified amount of times has a value 0 zero meaning forevever to stop agents automagically beling logged out by the system When agent picks up phone and is not on a local interface, a per-queue option to ask for confirmation by pressing a DTMF digit. This tells queue that call will be handled by this agent. %TC Can this option be a configuration of the agent.conf NOT queue.conf seems to me that it is the agent who would like that discretion not the queue process that should enforce this rule (I have already hacked this feature in chan_agent) If not confirmed, a per-queue option to log agent out or skip and place agent at bottom of queue. (Not really necessary but I could see it being useful for agents working from home with kids.) %TC again should this not be an agent.conf issue. also can we make sure the flag allows for 1-skip don't change agent place in Q
Re[2]: [Asterisk-Users] Asterisk PBX Billing
i think that the problem could be something more easy: it is possible inside asterisk to log all che calls of all the users and know the timing and the number called for each call? if it is possible to do that, could be possible to make a program that takes this files and generate the costs reading the log informations... so for me the real question is: there is a log of all the phone call that are made by asterisk? Angelo Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. It's at SS least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a pre-paid SS phone card system, where the shop prepays an overall amount for SS international calling access. Then you have to time each call as it is SS occurring, debiting each account, and the master account, in real-time. This SS can be a bit complex when you have 20 or 30 calls going at one time. You SS have to cut them off promptly when the money runs out (big problem). And SS you have to provide call detail and charges to them at the end of each call, SS using their own retail tariff. SS To add to the complexity, each country has a different tariff from the long SS distance carrier, and within the country, major cities often have special SS rates per minute. Mobiles have a different tariff too. Phone card SS platforms usually include a least-cost routing system which chooses a SS carrier real time based on the call. Tariffs change weekly and must be SS updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics, plasticity, and form. The greatest scientists are always artists as well. Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seg Fault!!
We had the same problem, we fixed it downgrading the Capi version to 0.2.1b Salut, -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de WipeOut . Enviado el: miércoles, 02 de julio de 2003 14:22 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Seg Fault!! Hi, I have just updated to the latest CVS version ( Approx 13:05 GMT Today ).. I am running chan_capi 0.2.2.. When a call is received Asterisk seg faults.. Not sure what information would be usefull to post so let me know what info will help to debug the problem.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asteri sk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re[2]: [Asterisk-Users] Asterisk PBX Billing
That's all I would need, it would be easy enough to work out the cost after that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angelo Sampietro Sent: Wednesday, July 02, 2003 10:06 AM To: Scott Stingel Cc: [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing i think that the problem could be something more easy: it is possible inside asterisk to log all che calls of all the users and know the timing and the number called for each call? if it is possible to do that, could be possible to make a program that takes this files and generate the costs reading the log informations... so for me the real question is: there is a log of all the phone call that are made by asterisk? Angelo Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. SS It's at least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a SS pre-paid phone card system, where the shop prepays an overall amount SS for international calling access. Then you have to time each call SS as it is occurring, debiting each account, and the master account, SS in real-time. This can be a bit complex when you have 20 or 30 calls SS going at one time. You have to cut them off promptly when the money SS runs out (big problem). And you have to provide call detail and SS charges to them at the end of each call, using their own retail SS tariff. SS To add to the complexity, each country has a different tariff from SS the long distance carrier, and within the country, major cities SS often have special rates per minute. Mobiles have a different SS tariff too. Phone card platforms usually include a least-cost SS routing system which chooses a carrier real time based on the call. SS Tariffs change weekly and must be updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I SS can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics, plasticity, and form. The greatest scientists are always artists as well. Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re[2]: [Asterisk-Users] Asterisk PBX Billing
There is a CDR (Call Detail Record) which is accessible in two different ways. The first is via a simple comma delimited file which can be parsed and fed into whatever database that you want. The second way is to dump the CDR directly into MySQL, and extract accordingly. So the only trick there is to create a database for billing and create a relationship that will extract from the CDR database. Kim C. Callis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angelo Sampietro Sent: Wednesday, July 02, 2003 7:06 AM To: Scott Stingel Cc: [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing i think that the problem could be something more easy: it is possible inside asterisk to log all che calls of all the users and know the timing and the number called for each call? if it is possible to do that, could be possible to make a program that takes this files and generate the costs reading the log informations... so for me the real question is: there is a log of all the phone call that are made by asterisk? Angelo Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. It's at SS least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a pre-paid SS phone card system, where the shop prepays an overall amount for SS international calling access. Then you have to time each call as it is SS occurring, debiting each account, and the master account, in real-time. This SS can be a bit complex when you have 20 or 30 calls going at one time. You SS have to cut them off promptly when the money runs out (big problem). And SS you have to provide call detail and charges to them at the end of each call, SS using their own retail tariff. SS To add to the complexity, each country has a different tariff from the long SS distance carrier, and within the country, major cities often have special SS rates per minute. Mobiles have a different tariff too. Phone card SS platforms usually include a least-cost routing system which chooses a SS carrier real time based on the call. Tariffs change weekly and must be SS updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics, plasticity, and form. The greatest scientists are always artists as well. Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Seg Fault!!
We had the same problem, we fixed it downgrading the Capi version to 0.2.1b what's the diff between 0.2.1b and 0.2.2 ? regards Marian Salut, -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de WipeOut . Enviado el: mircoles, 02 de julio de 2003 14:22 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Seg Fault!! Hi, I have just updated to the latest CVS version ( Approx 13:05 GMT Today ).. I am running chan_capi 0.2.2.. When a call is received Asterisk seg faults.. Not sure what information would be usefull to post so let me know what info will help to debug the problem.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asteri sk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linejack strikes again.
Hi All, Does anyone know if there has been any developments in asterisk to use LineJacks to dial out (connect to the PSTN)? The card works perfectly with virtually anything else but asterisk. Maybe the CVS versions have some work on it? Cheers, -Z -- __ Sign-up for your own FREE Personalized E-mail at Mail.com http://www.mail.com/?sr=signup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] Asterisk PBX Billing
thanks a lot! can you tell me where can i find more info about the CDR? probably this will be the better way to give to the company a summary with all the phone traffic :) Angelo Thursday, July 3, 2003, 4:37:32 PM, you wrote: KCC There is a CDR (Call Detail Record) which is accessible in two different KCC ways. The first is via a simple comma delimited file which can be parsed KCC and fed into whatever database that you want. The second way is to dump KCC the CDR directly into MySQL, and extract accordingly. So the only trick KCC there is to create a database for billing and create a relationship that KCC will extract from the CDR database. KCC Kim C. Callis KCC -Original Message- KCC From: [EMAIL PROTECTED] KCC [mailto:[EMAIL PROTECTED] On Behalf Of Angelo KCC Sampietro KCC Sent: Wednesday, July 02, 2003 7:06 AM KCC To: Scott Stingel KCC Cc: [EMAIL PROTECTED] KCC Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing KCC i think that the problem could be something more easy: KCC it is possible inside asterisk to log all che calls of all the users KCC and know the timing and the number called for each call? KCC if it is possible to do that, could be possible to make a program KCC that takes this files and generate the costs reading the log KCC informations... KCC so for me the real question is: there is a log of all the phone call KCC that are made by asterisk? KCC Angelo KCC Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. KCC It's at SS least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a KCC pre-paid SS phone card system, where the shop prepays an overall amount for SS international calling access. Then you have to time each call as it KCC is SS occurring, debiting each account, and the master account, in KCC real-time. This SS can be a bit complex when you have 20 or 30 calls going at one time. KCC You SS have to cut them off promptly when the money runs out (big problem). KCC And SS you have to provide call detail and charges to them at the end of KCC each call, SS using their own retail tariff. SS To add to the complexity, each country has a different tariff from KCC the long SS distance carrier, and within the country, major cities often have KCC special SS rates per minute. Mobiles have a different tariff too. Phone card SS platforms usually include a least-cost routing system which chooses KCC a SS carrier real time based on the call. Tariffs change weekly and must KCC be SS updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I KCC can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics, plasticity, and form. The greatest scientists are always artists as well. Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linejack strikes again.
Which driver are you using? Zara Trousk wrote: Hi All, Does anyone know if there has been any developments in asterisk to use LineJacks to dial out (connect to the PSTN)? The card works perfectly with virtually anything else but asterisk. Maybe the CVS versions have some work on it? Cheers, -Z ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA-186 de-register
Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I dont make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? Kim C. Callis
[Asterisk-Users] More switch = stuff
I have a two remote PBXs. I use the switch = statement on each PBX to point to the other PBX. Now I want extensions on PBX-1 to dial extensions and PSTN numbers that are local to PBX-2. However, I ALSO want people to be able to dial into a Zap channel on PBX-1 and be able to dial extensions on PBX-2, but NOT the PSTN numbers that are local to PBX-2. What would be the best way to do this? Do I have to set up separate users in iax.conf, one for using with a switch statement in the context calls to the Zap channel lands in and one user for calls from extensions land in? -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More switch = stuff
On Wed, 2003-07-02 at 10:36, Eric Wieling wrote: I have a two remote PBXs. I use the switch = statement on each PBX to point to the other PBX. Now I want extensions on PBX-1 to dial extensions and PSTN numbers that are local to PBX-2. However, I ALSO want people to be able to dial into a Zap channel on PBX-1 and be able to dial extensions on PBX-2, but NOT the PSTN numbers that are local to PBX-2. What would be the best way to do this? Do I have to set up separate users in iax.conf, one for using with a switch statement in the context calls to the Zap channel lands in and one user for calls from extensions land in? I believe you can specify contexts with the switch command. If so you can include a switch to an extensions context on the remote pbx in your inbound context while using a more full featured context for calls originating from inside. Look into that, if I am, then yes, you will need to create the second IAX user in each direction. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linejack strikes again.
I've tried all versions (stable CVS), even the latest new generation one (NIXJ) but no luck dialing out. All I can do is receive calls from the PSTN with it, but not making calls. Can you dial out with a linejack? Can you tell me how? Cheers, -Z - Original Message - From: Bruce Ferrell [EMAIL PROTECTED] Date: Wed, 02 Jul 2003 08:21:28 -0700 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Linejack strikes again. Which driver are you using? Zara Trousk wrote: Hi All, Does anyone know if there has been any developments in asterisk to use LineJacks to dial out (connect to the PSTN)? The card works perfectly with virtually anything else but asterisk. Maybe the CVS versions have some work on it? Cheers, -Z ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ Sign-up for your own FREE Personalized E-mail at Mail.com http://www.mail.com/?sr=signup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More switch = stuff
If you have to set up different users for the different contexts what' the usefulness of having a /context on the switch = statement? On Wed, 2003-07-02 at 10:47, Steven Critchfield wrote: On Wed, 2003-07-02 at 10:36, Eric Wieling wrote: I have a two remote PBXs. I use the switch = statement on each PBX to point to the other PBX. Now I want extensions on PBX-1 to dial extensions and PSTN numbers that are local to PBX-2. However, I ALSO want people to be able to dial into a Zap channel on PBX-1 and be able to dial extensions on PBX-2, but NOT the PSTN numbers that are local to PBX-2. What would be the best way to do this? Do I have to set up separate users in iax.conf, one for using with a switch statement in the context calls to the Zap channel lands in and one user for calls from extensions land in? I believe you can specify contexts with the switch command. If so you can include a switch to an extensions context on the remote pbx in your inbound context while using a more full featured context for calls originating from inside. Look into that, if I am, then yes, you will need to create the second IAX user in each direction. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re[4]: [Asterisk-Users] Asterisk PBX Billing
The mysql schema is available in the doc/cdr_mysql.txt file (from the asterisk source dir) James On Thu, 3 Jul 2003, Kim C. Callis wrote: You can find the comma delimited file at /var/log/asterisk/cdr-csv or if you are looking to do some easy querying on a database, you need to create a schema that I am sure someone on the channel has defined somewhere. At that point you clean up the /etc/asterisk/cdr_mysql.conf file to point to the appropriate database and authentication information. Kim C. Callis -Original Message- From: Angelo Sampietro [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 8:07 AM To: Kim C. Callis Cc: [EMAIL PROTECTED] Subject: Re[4]: [Asterisk-Users] Asterisk PBX Billing thanks a lot! can you tell me where can i find more info about the CDR? probably this will be the better way to give to the company a summary with all the phone traffic :) Angelo Thursday, July 3, 2003, 4:37:32 PM, you wrote: KCC There is a CDR (Call Detail Record) which is accessible in two different KCC ways. The first is via a simple comma delimited file which can be parsed KCC and fed into whatever database that you want. The second way is to dump KCC the CDR directly into MySQL, and extract accordingly. So the only trick KCC there is to create a database for billing and create a relationship that KCC will extract from the CDR database. KCC Kim C. Callis KCC -Original Message- KCC From: [EMAIL PROTECTED] KCC [mailto:[EMAIL PROTECTED] On Behalf Of Angelo KCC Sampietro KCC Sent: Wednesday, July 02, 2003 7:06 AM KCC To: Scott Stingel KCC Cc: [EMAIL PROTECTED] KCC Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing KCC i think that the problem could be something more easy: KCC it is possible inside asterisk to log all che calls of all the users KCC and know the timing and the number called for each call? KCC if it is possible to do that, could be possible to make a program KCC that takes this files and generate the costs reading the log KCC informations... KCC so for me the real question is: there is a log of all the phone call KCC that are made by asterisk? KCC Angelo KCC Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. KCC It's at SS least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a KCC pre-paid SS phone card system, where the shop prepays an overall amount for SS international calling access. Then you have to time each call as it KCC is SS occurring, debiting each account, and the master account, in KCC real-time. This SS can be a bit complex when you have 20 or 30 calls going at one time. KCC You SS have to cut them off promptly when the money runs out (big problem). KCC And SS you have to provide call detail and charges to them at the end of KCC each call, SS using their own retail tariff. SS To add to the complexity, each country has a different tariff from KCC the long SS distance carrier, and within the country, major cities often have KCC special SS rates per minute. Mobiles have a different tariff too. Phone card SS platforms usually include a least-cost routing system which chooses KCC a SS carrier real time based on the call. Tariffs change weekly and must KCC be SS updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I KCC can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics,
Re: [Asterisk-Users] Linejack strikes again.
No, u CANT dial out with a linejack... I have 2 of them, I use them as incoming only lines, and x100p for incoming, outgoing lines... There is no bug about this, is a feature, that isn't present at the linejack. On Wed, 2 Jul 2003, Zara Trousk wrote: I've tried all versions (stable CVS), even the latest new generation one (NIXJ) but no luck dialing out. All I can do is receive calls from the PSTN with it, but not making calls. Can you dial out with a linejack? Can you tell me how? Cheers, -Z - Original Message - From: Bruce Ferrell [EMAIL PROTECTED] Date: Wed, 02 Jul 2003 08:21:28 -0700 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Linejack strikes again. Which driver are you using? Zara Trousk wrote: Hi All, Does anyone know if there has been any developments in asterisk to use LineJacks to dial out (connect to the PSTN)? The card works perfectly with virtually anything else but asterisk. Maybe the CVS versions have some work on it? Cheers, -Z ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ Sign-up for your own FREE Personalized E-mail at Mail.com http://www.mail.com/?sr=signup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More switch = stuff
Sorry, that last sentence I had put in there was a thought interupted by my normal job. I meant for you to do the little research into adding the context to the end of your switch statement, and if I was wrong then go about the adding of users. On Wed, 2003-07-02 at 11:02, Eric Wieling wrote: If you have to set up different users for the different contexts what' the usefulness of having a /context on the switch = statement? On Wed, 2003-07-02 at 10:47, Steven Critchfield wrote: On Wed, 2003-07-02 at 10:36, Eric Wieling wrote: I have a two remote PBXs. I use the switch = statement on each PBX to point to the other PBX. Now I want extensions on PBX-1 to dial extensions and PSTN numbers that are local to PBX-2. However, I ALSO want people to be able to dial into a Zap channel on PBX-1 and be able to dial extensions on PBX-2, but NOT the PSTN numbers that are local to PBX-2. What would be the best way to do this? Do I have to set up separate users in iax.conf, one for using with a switch statement in the context calls to the Zap channel lands in and one user for calls from extensions land in? I believe you can specify contexts with the switch command. If so you can include a switch to an extensions context on the remote pbx in your inbound context while using a more full featured context for calls originating from inside. Look into that, if I am, then yes, you will need to create the second IAX user in each direction. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Enhanced queue app
or the agent sets a 'disposition' for that call, before it will exit wrapup and place back in queue. Jim Friedeck wrote: Wrap-up, as our existing phone system calls it, is a period of time that an agent will not get an incoming call after hanging up the previous call. This allows time for the agent to 'wrap-up' the preceeding conversation by filling out forms, typing on the computer, or taking a sip of coffee. Most systems have a variable wrap-up time for different agents or queues and a way for the agent to indicate he/she is immediately ready to take a call (like a programmable button on a digital phone.) Since Asterisk deals with analog phones, there is no programmable button we can use. Jim Friedeck -- Anton Yurchenko wrote: Benjamin Miller wrote: Would it be more flexible to approach this differently, with a dtmf to indicate that the agent is done with wrap up? So they get off a call and can wrap up the call for as long as necessary, and then hit * or something that marks them as available again rather than working against a timer to get a call wrapped up in 30 seconds or something, or shorting the timer because its always to long or something? Just my $.02 on the topic. Ben I`m looking forward to using * as a call center solution in future, and really excited about this thing. BUt I`m ashamed to say tham I`m not familuar with the terms, and what do wrap-up mean? -Original Message- From: Jim Friedeck [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 3:52 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Enhanced queue app Mark, How hard would it be to write a simple app to cancel wrap-up via an extension? Like dialing *99 to short- out the remaining wrap-up time? Jim Friedeck --- Mark Spencer wrote: Should wrap up time be something associated with a queue, or with an agent? Mark On Tue, 1 Jul 2003, Jim Friedeck wrote: Will try to change to this: Agent picks up phone and dials extension to 'login app': exten = 750,1,QueueLogin(QueueName, wrap-up-time) This would allow for quick agents to log into a queue for faster processing and allow slower processing for slow agents. An agent would simply log out if more time was needed. I could not think of a quick way to cancel wrap-up waiting. Our Inter-Tel has a programmable wrap-up cancel button. I don't think this would be very easy on POTS phones. Any ideas? Jim Friedeck -- TC wrote: I have also contracted mark for some minor modifications to app_queue and chan_agent 1) if you use a mixed environment of agents and devices on a single Q I want the ring process terminated before the time out value in queue.conf if the call is picked up by anyone assigned to the Q (device or agent) 2) if all agents are online when a new call comes into a Q, the current q logic will ring the devices for the timeout period before cycling and attempting to assign the called to a logged in agent , I want the Q to attempt to assign a call to an agent as soon as they hit the * key to hang up on the current call even if the ring process has started on the devices I also have some in line comments here see %TC To all who need more queue functionality, We are contracting Digium to enhance the queue app for our call center needs. Please read the following email conversation and give your ideas. Unless a glaring omission is found in my specification we will have them start tomorrow (Wednesday). I may not have thought of something important. It will be released to all Asterisk users by Digium. Thanks for your time. %TC THANK YOU JIM If agent recieves call while logged in and call goes unanswered for a specified amount of times (specified per queue) agent is logged out and event is recorded in CDR. Notification through astman interface would be desireable as well for management purposes. %TC Can we just make sure that specified amount of times has a value 0 zero meaning forevever to stop agents automagically beling logged out by the system When agent picks up phone and is not on a local interface, a per-queue option to ask for confirmation by pressing a DTMF digit. This tells queue that call will be handled by this agent. %TC Can this option be a configuration of the agent.conf NOT queue.conf seems to me that it is the agent who would like that discretion not the queue process that should enforce this rule (I have already hacked this feature in chan_agent) If not confirmed, a per-queue option to log agent out or skip and place agent at bottom of queue. (Not really necessary
[Asterisk-Users] Dialout Lines ???
I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call dropping
Are the 2 SIP UA's configured for the same codec? Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:05 AM Subject: [Asterisk-Users] Sip call dropping I'm having an issue with a connection between two sip phones, specfically sjphone, The two phones connect just fine, but after about 5 sec the call is dropped. On a side note a call does'nt got dropped between sip/sjphone and the outside line with a wx100p card. The communcation is on a full 100mbit network. I have a text file of the debug output of the call. Kevin, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 de-register
Yes, but I have been able to mitigate it by setting the following parameters. I have the problem with ATA's that are behind firewalls and not, but mostly with the ones that are behind firewalls. CfgInterval:1800 SIPRegInterval:100 On Thu, 3 Jul 2003, Kim C. Callis wrote: Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I don't make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BIG problem with multiple rings before pickup
Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
On Wed, 2003-07-02 at 13:34, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones and forward it to your fax machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip call dropping
Sjphone is set for Remote preferences for Codec Preference Selection Do you I want it Local preferences? Kevin, -Original Message- From: Michael Kane [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 12:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip call dropping Are the 2 SIP UA's configured for the same codec? Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:05 AM Subject: [Asterisk-Users] Sip call dropping I'm having an issue with a connection between two sip phones, specfically sjphone, The two phones connect just fine, but after about 5 sec the call is dropped. On a side note a call does'nt got dropped between sip/sjphone and the outside line with a wx100p card. The communcation is on a full 100mbit network. I have a text file of the debug output of the call. Kevin, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
On Wednesday 02 July 2003 01:34 pm, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. show application AbsoluteTimeout -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA-186 de-register
Same trouble J Regards Humberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kim C. Callis Sent: Jueves, 03 de Julio de 2003 10:34 a.m. To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA-186 de-register Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I dont make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? Kim C. Callis
RE: [Asterisk-Users] BIG problem with multiple rings before pickup
How do you tell asterisk to detect for fax tones? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, July 02, 2003 2:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BIG problem with multiple rings before pickup On Wed, 2003-07-02 at 13:34, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones and forward it to your fax machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
On 2003-07-02 at 13:54, Steven Critchfield ([EMAIL PROTECTED]) wrote: Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones Unfortunately, not all fax machines send the CNG tone, so using a separate fax number with distinctive ring is far more reliable. Can asterisk detect the various rings and route accordingly? If not, he could get two TDM100B cards, plug the two outputs of his distinctive ring detector into the two cards and route each one to a different context. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BIG problem with multiple rings before pickup
When the line is picked up, and a prompt is being played for the caller, asterisk listens for the beep, pause, beep, noise that an originating fax makes. When asterisk detects this it jumps to a fax extension in the current context to complete the call. This fax extension can be a dial string to redirect the call to the appropriate line. On Wed, 2003-07-02 at 14:12, Joe Antkowiak wrote: How do you tell asterisk to detect for fax tones? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, July 02, 2003 2:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BIG problem with multiple rings before pickup On Wed, 2003-07-02 at 13:34, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones and forward it to your fax machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BIG problem with multiple rings before pickup
You must put and fax exten in your context: For example: [default] ... exten = fax,1,Dial(Zap/2|30|d) srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Joe Antkowiak Enviado el: miércoles, 02 de julio de 2003 21:13 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] BIG problem with multiple rings before pickup How do you tell asterisk to detect for fax tones? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, July 02, 2003 2:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BIG problem with multiple rings before pickup On Wed, 2003-07-02 at 13:34, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones and forward it to your fax machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Thanks, but this is not a great solution. It still leaves the line off hook for the length of the timeout and limits real calls to the timeout as well. --On Wednesday, July 02, 2003 2:11 PM -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 02 July 2003 01:34 pm, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. show application AbsoluteTimeout -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BIG problem with multiple rings before pickup
You Answer on analog channels and then you need to have a fax extension in the current context. regards Martin On Wed, 2 Jul 2003, Joe Antkowiak wrote: How do you tell asterisk to detect for fax tones? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, July 02, 2003 2:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BIG problem with multiple rings before pickup On Wed, 2003-07-02 at 13:34, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones and forward it to your fax machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Hi Jim, You're probably not receiving disconnect-supervision on your analog lines, or have Zaptel configured incorrectly to recognize it. Check the list-archives (available from www.asterisk.org). You could try the busydetect-statement in zapata.conf. Also check Asterisk's main Makefile for some options related to busydetect. I strongly recommend the improved busydetect-routines (BUSYDETECT_MARTIN), which are not the default yet. Make sure that your voice-menu's always have a timeout, so it's impossible for your system to get stuck in one in the first place, should a caller not hang up or this fact not be detected by Asterisk. Grtz, Oliver On Wed, Jul 02, 2003 at 14:34:56 -0400, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BIG problem with multiple rings before pickup
On Wed, 2003-07-02 at 13:12, Joe Antkowiak wrote:
How do you tell asterisk to detect for fax tones?
create and exten for fax
exten = fax,1,Dial(${MYFAXDEVICE})
--Karl
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, July 02, 2003 2:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] BIG problem with multiple rings before pickup
On Wed, 2003-07-02 at 13:34, Jim Archer wrote:
Hi All...
I have a maddening problem...
I have Asterisk configured to pick up a line after 4 rings. I do this to
allow my fax machine to pick up a particular distinctive ring pattern, so
I
don't have to pay for a dedicated fax line.
If someone calls the line, lets it ring 3 times and then hangs up,
Asterisk
answers the line, and holds it off hook forever, constantly playing the
prompts.
My hardware is 2 X100P cards.
Any ideas?
Get a TDM10B, cancel your distinctive ring, and let asterisk answer
immediately and detect fax tones and forward it to your fax machine.
--
Karl Putland [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
On Wednesday 02 July 2003 02:42 pm, Jim Archer wrote: Thanks, but this is not a great solution. It still leaves the line off hook for the length of the timeout and limits real calls to the timeout as well. Not so. Once the user presses any DTMF, the first thing you can do is to set AbsoluteTimeout(0), which turns off the AbsoluteTimeout. If you originally set it to 60, the line will remain off hook for a maximum of 60 seconds before it will hang up. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
At 22:10 2003-07-01 -0400, you wrote: To find out what version yuor using, dial *999 and a debug/trace window will appear. In the SIP messages it will indicate the type of UA your using and the version. example below: try another call attempt with this window open and capture the call flow and send it to me. See below in bold or (User-Agent: X-PRO build 1035). Mike Mike, Thanks for writing. After you wrote, I had a brainwave and remembered that the UA is in the debug logs on asterisk... Be that as it may, I'm using X-Lite, and the latest version from their web site is build 1016. That's as of today (Wednesday, 07-02), and that's the version that I'm using on my system. I will send you the debug logs off-list, but here's the question: is it likely that X-Pro works but X-Lite still has trouble with its SDP? Regards, Moshe - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 9:32 PM Subject: Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite At 20:08 2003-07-01 -0400, Michael Kane wrote: What version of X-Lite are you using. The latest is build v1035. There where problems in earlier releases with SDP values, that could be the reason you not seeing invites or media. I had issues only with the media not setting as X-lite tried to negotiate media with another endpoint and teh SDP was hosed. Mike Mike, I downloaded the version I'm using late last week or early this week. It ought to be the latest. There's no way to tell from looking at the app (that I can find) what build it is. Let's see... created June 18th. That's pretty recent. I think it may be bug-report time. Any softphones you recommend for PC or for Linux? I'm actually rather disappointed with everything I've tested, with the exception of SJPhone (but I've only fiddled with it very briefly). Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 7:24 PM Subject: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite Today's frustrated programmer award goes to Linphone, which has the following debug output: (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that this is a linphone problem... other clients don't report problems. In other news, according to my trace of Ethernet packets, the PC softphone X-Lite sends no RTP packets -- neither UDP nor TCP -- to my Linux softphone, nor does it play out the UDP packets that it receives. This is not an asterisk problem because the PC's SJphone does work -- sortof. The PC's SJPhone does send/receive packets directly to asterisk. But there seems to be a problem with someone's negotiation protocol -- Kphone seems to expect GSM and SJPhone is apparently sending G.711. You can imagine how that sounds. More later if I get it straightened out. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Ah! I see, thanks! And thanks to everyone else. I have plenty to ways to proceed now. I'm going to try the cheap solutions first. Jim --On Wednesday, July 02, 2003 3:34 PM -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 02 July 2003 02:42 pm, Jim Archer wrote: Thanks, but this is not a great solution. It still leaves the line off hook for the length of the timeout and limits real calls to the timeout as well. Not so. Once the user presses any DTMF, the first thing you can do is to set AbsoluteTimeout(0), which turns off the AbsoluteTimeout. If you originally set it to 60, the line will remain off hook for a maximum of 60 seconds before it will hang up. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A solution for SIP and NAT
Hey Jim , you are correct in respect to the Service provider must pay for the bandwidth as I/we will be hair pinning calls back into the Internet. As far as voice quality is concerned (which is my biggest concern) the solution(box) FWD uses will not be the solution I will implement. I am seriously looking/talking with another vendor. But, back to the point, unless a SP has a subscriber base that is technically savy we/I really have no other choice. Again I have a STUN server implemented, but, SNOM and Grandstream are the only hardphone vendors that ship their products with a stun client(that I know of). Beleive me, I wish this wasn't something I had to think about. regards Mike Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 4:17 PM Subject: Re: [Asterisk-Users] A solution for SIP and NAT - Original Message - From: Michael Kane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 7:37 AM Subject: Re: [Asterisk-Users] A solution for SIP and NAT snip That why I have looked into(implemented) such technologies like STUN and probably will be forced to purchase a SIP aware firewall that will spoof and re-arrange SIP messages destined for my proxy server. snip Correct me if I am wrong but I see a couple big disadvantages to this solution. 1. Voice latency can be significantly increased since all the RTP traffic has to go through the VOIP providers NAT-proxy. Even if you are calling your next door neighbor, the traffic has to go all the way to the NAT-proxy and back. Just ask one of the FWD NAT-proxy users in Europe what it does for sound quality. 2. The VOIP provider has to pay for all the bandwidth of the RTP steams rather than just the small amount of traffic for call setup. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Erik over at XTEN is pretty good at getting the latest build out for both lite and prolet me email my partner they(xten just emailed us the latest and greatest for both. I also thought as of a week or so ago 1035 was the last build they would publish until their new product offering. (which is really cool) I'll email you lite if it's more recent than 1016.. Michael Kane (President/CEO) To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 4:43 PM Subject: Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite At 22:10 2003-07-01 -0400, you wrote: To find out what version yuor using, dial *999 and a debug/trace window will appear. In the SIP messages it will indicate the type of UA your using and the version. example below: try another call attempt with this window open and capture the call flow and send it to me. See below in bold or (User-Agent: X-PRO build 1035). Mike Mike, Thanks for writing. After you wrote, I had a brainwave and remembered that the UA is in the debug logs on asterisk... Be that as it may, I'm using X-Lite, and the latest version from their web site is build 1016. That's as of today (Wednesday, 07-02), and that's the version that I'm using on my system. I will send you the debug logs off-list, but here's the question: is it likely that X-Pro works but X-Lite still has trouble with its SDP? Regards, Moshe - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 9:32 PM Subject: Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite At 20:08 2003-07-01 -0400, Michael Kane wrote: What version of X-Lite are you using. The latest is build v1035. There where problems in earlier releases with SDP values, that could be the reason you not seeing invites or media. I had issues only with the media not setting as X-lite tried to negotiate media with another endpoint and teh SDP was hosed. Mike Mike, I downloaded the version I'm using late last week or early this week. It ought to be the latest. There's no way to tell from looking at the app (that I can find) what build it is. Let's see... created June 18th. That's pretty recent. I think it may be bug-report time. Any softphones you recommend for PC or for Linux? I'm actually rather disappointed with everything I've tested, with the exception of SJPhone (but I've only fiddled with it very briefly). Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 7:24 PM Subject: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite Today's frustrated programmer award goes to Linphone, which has the following debug output: (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that this is a linphone problem... other clients don't report problems. In other news, according to my trace of Ethernet packets, the PC softphone X-Lite sends no RTP packets -- neither UDP nor TCP -- to my Linux softphone, nor does it play out the UDP packets that it receives. This is not an asterisk problem because the PC's SJphone does work -- sortof. The PC's SJPhone does send/receive packets directly to asterisk. But there seems to be a problem with someone's negotiation protocol -- Kphone seems to expect GSM and SJPhone is apparently sending G.711. You can imagine how that sounds. More later if I get it straightened out. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
your right.1016 is the latest and greatest... Michael Kane (President/CEO) To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 4:43 PM Subject: Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite At 22:10 2003-07-01 -0400, you wrote: To find out what version yuor using, dial *999 and a debug/trace window will appear. In the SIP messages it will indicate the type of UA your using and the version. example below: try another call attempt with this window open and capture the call flow and send it to me. See below in bold or (User-Agent: X-PRO build 1035). Mike Mike, Thanks for writing. After you wrote, I had a brainwave and remembered that the UA is in the debug logs on asterisk... Be that as it may, I'm using X-Lite, and the latest version from their web site is build 1016. That's as of today (Wednesday, 07-02), and that's the version that I'm using on my system. I will send you the debug logs off-list, but here's the question: is it likely that X-Pro works but X-Lite still has trouble with its SDP? Regards, Moshe - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 9:32 PM Subject: Re: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite At 20:08 2003-07-01 -0400, Michael Kane wrote: What version of X-Lite are you using. The latest is build v1035. There where problems in earlier releases with SDP values, that could be the reason you not seeing invites or media. I had issues only with the media not setting as X-lite tried to negotiate media with another endpoint and teh SDP was hosed. Mike Mike, I downloaded the version I'm using late last week or early this week. It ought to be the latest. There's no way to tell from looking at the app (that I can find) what build it is. Let's see... created June 18th. That's pretty recent. I think it may be bug-report time. Any softphones you recommend for PC or for Linux? I'm actually rather disappointed with everything I've tested, with the exception of SJPhone (but I've only fiddled with it very briefly). Michael Kane To-Talk Communications LLC. 37 Sandusky Dr. Wareham, Ma. 02571 www.to-talk.com 508-295-2826 - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 01, 2003 7:24 PM Subject: [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite Today's frustrated programmer award goes to Linphone, which has the following debug output: (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that this is a linphone problem... other clients don't report problems. In other news, according to my trace of Ethernet packets, the PC softphone X-Lite sends no RTP packets -- neither UDP nor TCP -- to my Linux softphone, nor does it play out the UDP packets that it receives. This is not an asterisk problem because the PC's SJphone does work -- sortof. The PC's SJPhone does send/receive packets directly to asterisk. But there seems to be a problem with someone's negotiation protocol -- Kphone seems to expect GSM and SJPhone is apparently sending G.711. You can imagine how that sounds. More later if I get it straightened out. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Ok, how do I detect the pressing of any touch tone so I can set the timeout back to 0? --On Wednesday, July 02, 2003 2:51 PM -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 02 July 2003 02:12 pm, Joe Antkowiak wrote: How do you tell asterisk to detect for fax tones? Zaptel devices will detect fax tones automatically. If it finds them, Asterisk will attempt to go to extension fax, priority 1, if it exists. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Actually there is a WaitForRing app that would probably solve this more easily. Mark On 2 Jul 2003, Steven Critchfield wrote: On Wed, 2003-07-02 at 13:34, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones and forward it to your fax machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
basically make priority 1 of any extension other than timeout be a set timeout 0. On Wed, 2003-07-02 at 16:32, Jim Archer wrote: Ok, how do I detect the pressing of any touch tone so I can set the timeout back to 0? --On Wednesday, July 02, 2003 2:51 PM -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 02 July 2003 02:12 pm, Joe Antkowiak wrote: How do you tell asterisk to detect for fax tones? Zaptel devices will detect fax tones automatically. If it finds them, Asterisk will attempt to go to extension fax, priority 1, if it exists. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
On Wednesday 02 July 2003 04:32 pm, Jim Archer wrote: Ok, how do I detect the pressing of any touch tone so I can set the timeout back to 0? Pressing DTMF while waiting for a timeout (or while playing a file with Background) will redirect you to a new extension. For example, if you call us (700-382-4758), you'll hear a looping greeting. If you type in my extension (103), it'll jump to extension 103, priority 1, where I run AbsoluteTimeout(0), then it runs Dial on my extension. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Mark, I took a look at this app. How would I do this? Put wait for ring at the top of the loop? How do I detect and act on the return value? --On Wednesday, July 02, 2003 4:38 PM -0500 Mark Spencer [EMAIL PROTECTED] wrote: Actually there is a WaitForRing app that would probably solve this more easily. Mark On 2 Jul 2003, Steven Critchfield wrote: On Wed, 2003-07-02 at 13:34, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones and forward it to your fax machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Ok, thanks. I was hoping I would not have to set timeout back to 0 for each extension... --On Wednesday, July 02, 2003 4:54 PM -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 02 July 2003 04:32 pm, Jim Archer wrote: Ok, how do I detect the pressing of any touch tone so I can set the timeout back to 0? Pressing DTMF while waiting for a timeout (or while playing a file with Background) will redirect you to a new extension. For example, if you call us (700-382-4758), you'll hear a looping greeting. If you type in my extension (103), it'll jump to extension 103, priority 1, where I run AbsoluteTimeout(0), then it runs Dial on my extension. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] client reinvitation problem
Hello All! There is description of my problem with Asteriks below. Asteriks CLI says: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call Sip debug on the server gives the next: Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.26:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=106403508 To: sip:[EMAIL PROTECTED];user=phone;tag=as0771c6f9 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 209 8523 is Cisco ATA-186 The sip.conf content: - - - - - [cisco8523] type=friend username=8523 secret=test nat=no host=dynamic canreinvite=no qualify=300 defaultip=192.168.0.26 - - - - - Why I place a call to Asteriks. I hear some invitation but connection brokes when retransmit exceed. Could anyone give some advice or solution. Thanks in advance -- Best regards Vlad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] ANNOUNCE: CLASS-like features for Asterisk
Yo all,
As there has been some intrest, here's my updated version:
I post it to -dev as well as -users, as it may be of intrest to
both.
Inspired by the example in the tips tricks-section of
http://www.junghanns.net/asterisk/;, I built a more elaborate
set of features. Currently, my implementation supports call-
forward unconditional, on no answer and on busy. It furthermore
provides each extension with a set of user-definable speed-dials.
It validates if the number entered is actually valid for the current context
and additionally saves this context into the DB and always uses it to
originate the divert from, as you can't expect the forward destination-
number to be available from all caller's contexts. Only the
speed-dials use the extension's current context, but do save the
context a speed-dial was entered from into the DB anyways.
Some things you need to know:
- No check for forwarding-loops yet. I haven't tried what happens
when you create one, either. :-)
- When a call gets forwarded to another local extension and goes to
voicemail from there, voicemail will be left for the extension that
originally forwarded the call, instead of the extension the call was
forwarded to. This might be seen as a feature.
- The contexts to set the features are to be included into the
context of each extension you want to allow to use them and you
should replace your macro-stdexten with my version (which accepts
some more args than the original version, to allow for passing most
arguments to the Dial-application. This means you have to change
either your extension-entries or the macro, if you're already using
macro-stdexten.
- If you where using my first version, you should remove all related
entries from your ast-DB, as I modified the format somewhat.
- As you can see, it currently uses Festival for it's prompts,
which should obviously be installed and working with Asterisk.
Without it, everything will probably still work, but no prompts,
except the standard invalid-recording, which is used when entering
invalid numbers, will be played.
- I'll probably add more features when I have time, and post an
update. Also, this stdexten-macro, like the original one,
assumes everyone has voicemail. Making a second copy without
voicemail or adding an extra argument to enable / disable VM-
processing should be trivial.
Usage:
921 + number - Set unconditional forwarding.
921 - Cancel unconditional forwarding.
9921 - Check unconditional forwarding.
961 + number - Set forwarding on no answer.
961 - Cancel forwarding on no answer.
9961 - Check forwarding on no answer.
967 + number - Set forwarding on busy.
967 - Cancel forwarding on busy.
9967 - Check forwarding on busy.
970 - 999 + number - Set a personal speed-dial.
970 - 999 + 0 - Clear a personal speed-dial.
970 - 999- Call a personal speed-dial, if set.
9970 - - Check a personal speed-dial.
Note that I choose 9 instead of * because a lot of IP-phones
don't allow the user to dial a number containing the * or #.
Nothing stops you from changing it as needed.
Also note that this is almost my first go at it, and I haven't tested it
very heavily. Comments, suggestions and additions are welcome. Hope
it's useful to some of you.
Grtz,
Oliver
;
; Macros
;
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - Timeout
; ${ARG4} - Other options to app_dial
;
exten = s,1,DBget(fwdexten=FEAT/${ARG1}/CFWD/CFU)
exten = s,102,Goto(s|4)
exten = s,2,DBget(fwdcontext=FEAT/${ARG1}/CFWD/CFUC)
exten = s,3,Goto(${fwdcontext}|${fwdexten}|1)
exten = s,4,Dial(${ARG2},${ARG3},${ARG4})
exten = s,105,Goto(s|205)
exten = s,5,DBget(fwdexten=FEAT/${ARG1}/CFWD/CFNA)
exten = s,106,Goto(s|8)
exten = s,6,DBget(fwdcontext=FEAT/${ARG1}/CFWD/CFNAC)
exten = s,7,Goto(${fwdcontext}|${fwdexten}|1)
exten = s,8,Answer
exten = s,9,Voicemail2(su${ARG1})
exten = s,10,Hangup
exten = s,205,DBget(fwdexten=FEAT/${ARG1}/CFWD/CFB)
exten = s,306,Goto(s|208)
exten = s,206,DBget(fwdcontext=FEAT/${ARG1}/CFWD/CFBC)
exten = s,207,Goto(${fwdcontext}|${fwdexten}|1)
exten = s,208,Answer
exten = s,209,Voicemail2(sb${ARG1})
exten = s,210,Hangup
;
; Special features, Call Forwarding, unconditional.
;
[feature-cfu]
exten = _921X.,1,Answer
exten = _921X.,2,ChanIsAvail(Local/${EXTEN:[EMAIL PROTECTED])
exten = _921X.,103,Playback(invalid)
exten = _921X.,104,Hangup
exten = _921X.,3,DBput(FEAT/${CALLERIDNUM}/CFWD/CFU=${EXTEN:3})
exten = _921X.,4,DBput(FEAT/${CALLERIDNUM}/CFWD/CFUC=${CONTEXT})
exten = _921X.,5,Festival(Call-Forward Unconditional: Has been set too: ${EXTEN:3}.)
exten = _921X.,6,Hangup
exten = 921,1,Answer
exten = 921,2,DBdel(FEAT/${CALLERIDNUM}/CFWD/CFU)
exten = 921,3,DBdel(FEAT/${CALLERIDNUM}/CFWD/CFUC)
exten =
[Asterisk-Users] Sorry 'bout that
Sorry 'bout that vacation message. Procmail usually is smart enough to avoid sending replies to mailing lists. I put in a rule to prevent this from happening again. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialout Lines ???
Yes you can. Configure it either as a SIP gateway or an h.323 gatekeeper. Bradley Greep wrote: I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
