Re: [asterisk-users] does any one knows of a Softphone that works under terminal services?
I wonder if I setup a softphone on each terminal if they will actually work as independent phones well enough, but haven´t tested it. MF escribió: Hi all I'm looking for a softphone that works well under terminal services environment, we need to set up 24 to 32 phones for a call center, also, does any one knows if it will actually work fine under load? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to detect long calls
I had the same problem last year, at the time for some reason Timeout
instruction wouldn't trigger, so, just to be sure not to have to pay
for another longdistance call, I did the following, (following
someone's advise in here)
/usr/sbin/asterisk -rx show channels concise |awk -F : '($11 1500)
{print /usr/sbin/asterisk -rx \soft hangup $1 \} '|sh
this will hangup any call longer than 1500 seconds, or what ever
value you choose
hope it helps you somehow
;-)
Manrique
Cullin J. Wible escribió:
You should:
Set(TIMEOUT(absolute)=14400)
When the call is received - this will set the maximum limit of a call and
asterisk will force hang-up when the limit is reached.
14400 seconds = 4 hours, which for our purposes is longer then any call we
expect. Even if you double-it or set it to several days some limit is better
then nothing.
When we found the same problem we had a call that was stuck open for 20
days. The call was stuck in a conference and was sending the on-hold music,
which is what kept it open.
Hope that helps.
Cullin J. Wible
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
Sent: Tuesday, January 16, 2007 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to detect long calls
Savoy, Kevin - Williston, ND wrote:
We have been running an Asterisk box with 1.2.9.1 on it since August
in a call center environment. We use the Asterisk box as an IVR and
then pass the calls on to a Nortel Option 11C. Today we found in our
long distance bill two calls that lasted a VERY long time. One was 58
hours and another was 38 DAYS!!!
Nortel does not show this call being that long. Obviously the person
that called in didn't hold the line for 58 days so somehow between
Asterisk and MCI the call got stuck open and didn't hang up on the
network.
My question is two parts, part one, has anyone heard of anything like
this where a call doesn't hang up properly and seems stuck in the
system. Part two is there anyway to monitor in Asterisk the length of
all active calls and then if a call lasts longer then, say one hour,
we could send off a text message or warning.
Hi ,
similiar thing happend to me. Try looking at the L() optin in Dial. I
define a max call time, say few hours, then warn every x seconds, then cut
the call.
--
thanks,
Yusuf
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R D Director
[EMAIL PROTECTED]
Kínetos Software
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408-538-2113
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[asterisk-users] How to notify an ACD agent before he/she picks up
Hi, I need to send a message to an agent when the ACD starts to ring on he/she. I have and application already built that sends such a message (just like a cti), just don't know how to get from asterisk which agent was selected prior to ringing him (or during ringing), so that I can get information about the call and send it over. any one done this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)
thanks CF, I did change the PRI CAUSE to unavailable, or reject. only that it still shows Accepting overlap call from. just before this -Executing SetVar(Zap/12-1, PRI_CAUSE=27) does anyone knows if this call being picked up at anytime? Problem is, this is a reverse charge line with more than 3000 calls per hour, and if it telco thinks it is picked up for a milisecond will charge for the whole minute. But I can't disconnect the service since it is needed during 2 hours a day on a TV show.(that's the only time when people should be calling, but they keep calling the whole day instead) C F escribió: Set the PRI cause: http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+PRI_CAUSE On 8/15/06, Manrique Feoli [EMAIL PROTECTED] wrote: Hi, I´m in a bit of a hurry here, I need to reject calls before picking them up. If I do hangup on the first line, does anyone knows if the line counts as picked up for the Telco? how about if I register the incoming callerid, and then do hangup on the second line? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)
Hi, I´m in a bit of a hurry here, I need to reject calls before picking them up. If I do hangup on the first line, does anyone knows if the line counts as picked up for the Telco? how about if I register the incoming callerid, and then do hangup on the second line? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN) In a Hurry
Basically, if I do hangup on the first line the console shows: Starting simple switch on Zap/2-1 Accepting overlap call from '' to '3423' on channel 0/1, span 1 executing Hangup (Zap/2-1, ) in new stack. I believe this is actually picking the call up isn't it? Manrique Feoli escribió: Hi, I´m in a bit of a hurry here, I need to reject calls before picking them up. If I do hangup on the first line, does anyone knows if the line counts as picked up for the Telco? how about if I register the incoming callerid, and then do hangup on the second line? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?
Hi all, I need to setup 6 phones about 3/4 of a mile from the main box, (can't do it with VoIP yet because of networking issues), does anyone knows if the boards can resist such a length for FXS ports. Right now there is a Dialogic MSI160 working fine. The actual length in a straight line is about half a mile, it does have a single cable point to point, but can't be sure where the cable goes since it has to pass from one building to another, so it might end up been quite a bit more distance. If the answer is they can't handle such a distance does anyone knows if there is an alternative, (line booster or whatever similar ) thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?
Thanks Rich, I was expecting that. I was worried because on other type of cards if you set the phone too far you'll burn the port of the card, mainly because of the lack of capacity to keep such a long line up. Now when you say 2 or three ringers, you mean 2 or three ring events or phones ringing at once or what did you mean? I need 6 phones now, and not expecting more than 2 ringing at once. (hopefully) cheers Manrique Rich Adamson escribió: Manrique Feoli wrote: Hi all, I need to setup 6 phones about 3/4 of a mile from the main box, (can't do it with VoIP yet because of networking issues), does anyone knows if the boards can resist such a length for FXS ports. Right now there is a Dialogic MSI160 working fine. The actual length in a straight line is about half a mile, it does have a single cable point to point, but can't be sure where the cable goes since it has to pass from one building to another, so it might end up been quite a bit more distance. If the answer is they can't handle such a distance does anyone knows if there is an alternative, (line booster or whatever similar ) With the exception of the ring generator on the digium card, the fxs port specs are basically the same as telephone company specs. Telco lines in rural areas often times exceed six miles. So, yes the digium card will handle your half mile just fine. The ring generator on the card is much smaller then those used in the telco, and is likely limited to maybe two or three ringers at the distant location. Don't plan on attaching multiple phones to the fxs port at your remote location. (Same is basically true with the sangoma card.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme chat room with many users, and only 4 can talk, is there a max amount of users?
Hi I need to setup a meetme room where you could accept say 120 incoming calls to listen to the chat, BUT, only the first 4 can talk, so when one of the first 4 leaves the room, number 5 becomes 4 and is able to talk on the room. Is this doable with meetme?? (am I making any sense?) Is this a reasonable amount of calls to handle with meetme?¿? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Meetme chat room with many users, and only 4 can talk, is there a max amount of users?
Thanks for your reply Tony(I'm starting a new thread now, sorry about that!!) I see we'll have to do some developing for what I'm looking for, but thinking again about it, I should be able to make one room for chatting of 4 users, and one room only for listening of about 120 calls, shouldn't I? Now, talking about capacity with meetme, If I work with a quad span E1, getting all connections to talk between them on meetme, (that is not transcoding and not connecting to SIP phones). Will this take too much CPU? Does anyone knows or thinks I might be able to manage 120 channels conferencing among them only on zap devices with a PIV/ 3GHz 1GB Ram? thanks Manrique I need to setup a meetme room where you could accept say 120 incoming calls to listen to the chat, BUT, only the first 4 can talk, so when one of the first 4 leaves the room, number 5 becomes 4 and is able to talk on the room. Is this doable with meetme?? (am I making any sense?) Not directly. You will need some kind of program that is using the Manager API to monitor activity and issue the unmute commands. Alternatively, make your own customised version of MeetMe itself. You should also consider how you might notify a caller when they become eligible to speak. Is this a reasonable amount of calls to handle with meetme?¿? That depends on many factors: CPU power; whether the calls come in via a quad-E1 card or VoIP; for VoIP, what codec is used; etc. Cheers Tony ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is the manager good for high traffic?? but only with one connection to it
Hi, I've read all over that the manager conection (via sockets) isn't good for high traffic applications with multiple manager connections at the same time with one asterisk, the connection hangs and many other problems. having said that, my question is: Has anyone worked on a fairly high traffic environment BUT with ONLY ONE connection to administer asterisk via the manager, that is to do Login/Logout call generation, etc??that is sending commands and receiving many events per minute or even per second. I'm talking about 60 lines / 2 E1 with working full at some peak times, with 30 agents on SIP. is it stable enough?what other way should I go if not. I appreaciate any point of view, or past experiences anything thanx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't retake call after dialing through Zap/E1 wich doesn't answer
I explained it backwards,
the thing is I need to make a call right when an event happens, for
example when the second link is down, or when I receive a particular call.
In the following sample, I get a call on the first span E1 (g1), and
transfer it to the second span (g0). IF the link is down, I would
like to call support and let them know.
problem is when line 2 has noanswer line 3 never gets executed.
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},5,r)(so if the second link
doen't answer after 5 seconds, it should play a message and call support)
exten = _X.,3,Dial(Zap/g0/${SUPPORT_PHONE},30,r)
exten = _X.,4,Playback(help)
this is another one, that can't make work with the same situation, I
can't hangup the call on the E1 slot without ending the call itself,
I've tested hangup and softhangup
exten =7595,1,answer
exten =7595,2,playback(hello)
exten =7595,3,softhangup(${channel}|a)
exten = 7595,4,Dial(Zap/g0/8734438,60,tr)
exten =7595,4,playback(muchasgracias)
exten =7595,5,hangup
All this to try to do it on the same context, (trying to avoid making a
call file ),
maybe it doesn't make any sense does it?
Manrique Feoli escribió:
Maybe the question is, how can I call someone right after I something
happens, in this particular case if the Dial is not answered.
Manrique Feoli escribió:
Hi all,
I am receiving a call on one E1 and try to set up a call on another
E1, if the second call succeds, fine but if the second call
doesn't answer (or if the second E1 link happens to be down)I
can't manage to execute another line of my dialplan to try to setup
the call via another route.
I must be missing something basic.
here are my dialplay lines (taken to the simplest expresion)
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},5,r)(so if the second link
doen't answer after 5 seconds, it should play a message and call
support)
exten = _X.,3,Playback(help)
exten = _X.,4,Dial(Zap/g0/${SUPPORT_PHONE},30,r)
Line 2 jumps to the h priority, and doesn't execute line 3.
any clue?
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Re: [asterisk-users] can't retake call after dialing through Zap/E1 wich doesn't answer
Maybe the question is, how can I call someone right after I something
happens, in this particular case if the Dial is not answered.
Manrique Feoli escribió:
Hi all,
I am receiving a call on one E1 and try to set up a call on another
E1, if the second call succeds, fine but if the second call
doesn't answer (or if the second E1 link happens to be down)I
can't manage to execute another line of my dialplan to try to setup
the call via another route.
I must be missing something basic.
here are my dialplay lines (taken to the simplest expresion)
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},5,r)(so if the second link
doen't answer after 5 seconds, it should play a message and call
support)
exten = _X.,3,Playback(help)
exten = _X.,4,Dial(Zap/g0/${SUPPORT_PHONE},30,r)
Line 2 jumps to the h priority, and doesn't execute line 3.
any clue?
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**
Manrique Feoli
Gerente Investigación y Desarrollo
[EMAIL PROTECTED]
Kínetos Telefonía e Informática.
www.kinetos.com
506-234-7771
**
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Re: [asterisk-users] Trunk transferring?
man what can I say, i've never heard of that, but if you find a way to transfer a call from an 800 line to a no charge trunk within the same call, I'll be very interested in learning it. regards Manrique David Freeman escribió: Hello all. I'm very new to the list and quite new to Asterisk. I did a little messing around with a prebuilt system (TrixBox) while investigating phone systems for a business that my partner and I are getting together. Anyway, what I'm wondering is if it's possible to do this: A call comes in to a US 800 number that is a SIP or IAX VoIP line, then * can transfer that incoming call to use another SIP or AIX VoIP line? I want to offer an 800 number (and pay for the initial time for the call) but I don't want to pay 800 charges for the duration of the call (we could be on the phone with someone an hour or longer.) I've seen references to trunk transferring in some commercial products for *, but I'm thinking if they have that functionality, I should be able to build it, too! Thanks in advance. Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ** Manrique Feoli Gerente Investigación y Desarrollo [EMAIL PROTECTED] Kínetos Telefonía e Informática. www.kinetos.com 506-234-7771 ** ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can't retake call after dialing through Zap/E1 wich doesn't answer
Hi all,
I am receiving a call on one E1 and try to set up a call on another E1,
if the second call succeds, fine but if the second call doesn't
answer (or if the second E1 link happens to be down)I can't manage
to execute another line of my dialplan to try to setup the call via
another route.
I must be missing something basic.
here are my dialplay lines (taken to the simplest expresion)
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},5,r)(so if the second link
doen't answer after 5 seconds, it should play a message and call support)
exten = _X.,3,Playback(help)
exten = _X.,4,Dial(Zap/g0/${SUPPORT_PHONE},30,r)
Line 2 jumps to the h priority, and doesn't execute line 3.
any clue?
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Re: [asterisk-users] long distance ethernet Asterisk
If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Two questions: We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 – 10Mbps. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long distance ethernet Asterisk
another thought, if you are in a bowl, all you need to find is line of sight to one common place from both ends, and place a repeater there. (you could also set two or three steps repeating the signal within points which have line of sight). I'm not sure but I think one repeater would be much cheaper than 20.000ft of copper + extenders + poles+ maintenance, lighning... (even thought you are in Copper Mountain !!, BTW nice spot ). if in the end you decide to go with ethernet, just beware of lighning!!! Brian Vincent (C) escribió: I know.. I know… fiber would be ideal. We have single-mode all over the place. We even have some dark, unterminated strands within 2000ft of this location – it makes me want to cry. Unfortunately lighting it up isn’t an option – we wouldn’t gain anything because we couldn’t connect to anything else to get us the last stretch. Trenching 2000ft isn’t an option – this is National Forest land and we’re not allowed to do that. As far as wireless – no line of sight. This location sits in a little bowl at 11,200’. So what I’m left with is a 400pr, 22awg out to 3000’. Then we jump on 200pr, 24awg aerial cable strung on the 3^rd longest high-speed quad chairlift (10,800’ run). The last leg involves a short underground to another high-speed quad and down 6000’. We can stick a powered repeater in the motor room of the first lift (so I guess a bit further than the original 12,000’ I was thinking.) Yes, we do strange things. If you’re really curious, here’s a map of the campus environment we maintain: http://www.skireport.com/colorado/copper/trailmap/ --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Bruce Reeves *Sent:* Thursday, July 27, 2006 4:03 PM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] long distance ethernet Asterisk I would really look towards fiber, the bandwidth and distance can easily be handled. On 7/27/06, *Manrique Feoli* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, *Brian Vincent (C)* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Two questions: 1. We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 – 10Mbps. 2. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] __ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. __ ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks
Re: [asterisk-users] Trunk transferring?
why not calling them back? 1- you get the call on the 800 line, and after the operator evaluaes the situation, dials a digit or something, the system calls back the same number, but via your preferred route/system/billing. 2- get the call, then play an automated message to the user explaining how and why we'll call him back in a minute, so he hangs up and wait for the call David Freeman escribió: Hello all. I'm very new to the list and quite new to Asterisk. I did a little messing around with a prebuilt system (TrixBox) while investigating phone systems for a business that my partner and I are getting together. Anyway, what I'm wondering is if it's possible to do this: A call comes in to a US 800 number that is a SIP or IAX VoIP line, then * can transfer that incoming call to use another SIP or AIX VoIP line? I want to offer an 800 number (and pay for the initial time for the call) but I don't want to pay 800 charges for the duration of the call (we could be on the phone with someone an hour or longer.) I've seen references to trunk transferring in some commercial products for *, but I'm thinking if they have that functionality, I should be able to build it, too! Thanks in advance. Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up.
Hey I need a quick advise here, I must be missing something basic.
I get a call from an Zap E1, and dial into a Voip extension,
if the extension hangs up first, the next line of the dialplan gets
executed,
if the pstn hangs up first, shows exited non-zero on ZAP/6-1 and
the next line doesn't get executed. ( 3,system(...) )
this is my dialplan
exten =_X.,1,Answer
exten =_X.,2,Dial(Zap/g1/${EXTEN},,tTrhH)
exten =_X.,3,system(/bin/echo -e '${DIALEDTIME}, ${ANSWEREDTIME},
${TIMESTAMP},${CALLERID},${EXTEN},${CHANNEL}' /home/mque/E1/list_calls)
exten =_X.,4,Congestion
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Re: [asterisk-users] Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up.
thanks for your worthy advise Andres, that in deed does the trick.
I had actually thought about that solution, but then I'll have to
evaluate all calls again at hangup ( h ) to see how to handle their
end,
That in my case wasnt all that nice given I need different types of
finishing funtions to be performed according to what the call went like
and to the type of call. Besides that it made the dial plan less readable.
I thought maybe if there was a way to avoid this exited non-zero on
ZAP/6-1 situation I could handle each finishing right at each
extensions end.
Andres escribió:
Manrique Feoli wrote:
Hey I need a quick advise here, I must be missing something basic.
I get a call from an Zap E1, and dial into a Voip extension, if the
extension hangs up first, the next line of the dialplan gets executed,
if the pstn hangs up first, shows exited non-zero on ZAP/6-1 and
the next line doesn't get executed. ( 3,system(...) )
this is my dialplan
exten =_X.,1,Answer
exten =_X.,2,Dial(Zap/g1/${EXTEN},,tTrhH)
exten =_X.,3,system(/bin/echo -e '${DIALEDTIME}, ${ANSWEREDTIME},
${TIMESTAMP},${CALLERID},${EXTEN},${CHANNEL}'
/home/mque/E1/list_calls)
You can try to put this in the 'h' extension so it gets executed upon
hangup:
exten = h,1,system(/bin/echo -e '${DIALEDTIME}, ${ANSWEREDTIME},
${TIMESTAMP},${CALLERID},${EXTEN},${CHANNEL}'
/home/mque/E1/list_calls)
exten =_X.,4,Congestion
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**
Manrique Feoli
Gerente Investigación y Desarrollo
[EMAIL PROTECTED]
Kínetos Telefonía e Informática.
www.kinetos.com
506-234-7771
**
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Re: [asterisk-users] Connecting branch offices through IPsec tunnel -- latency effects?
Ive done it with a tunnel set with OpenVPN, and works quite good, there is a slight increase of lattency but not noticeable to humans. that is doing it via UDP tunnel, we also tried via a TCP tunnel and results weren't good, lattency increased more than desired and voice quality was poor. other than that enough CPU is very advisable Stephen Bosch escribió: Hi: If I connect two offices through an IPsec tunnel, what is the impact on latency, and does it noticeably affect calls? Has anyone out there tried this? What were the effects? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)
Hi all, Here is the situation: A call comes in to an Alcatel PBX and it sends it to an E1 on * , this * either sends the call to a VoIP extension or needs to forward it to an extension back on the Alcatel, but WITHOUT using another slot of the E1 (no tromboning or hairpinning). I've read you can do this with 2b channel transfers implemented on 5ESS, and also on QSIG. I know Matthew Fredrickson did it on * (I think he programmed it for *) I also know there is quite a bit of people pursuing this same goal, which is way important to lower the income barriers for * to enter the legacy world. Has anyone actually done it? I appreciate any input whatsoever, and if possible a sample of how to manage it on *.What to put on the extensions.conf to perform the transfer and any other files needed, thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] transfers from an E1 using 2b-channel or similar anyone? (QSIG?)
Hi Matt, thanks for your answer, I guess it is still as you said a while back that you did it using 5ESS Can you share how you did in 5ESS? (a sample of the extensions.conf ) and what kind of switch you were connected to? I'm not sure if the Alcatel 4400 and the Nortel Meridian 11 supports 5ESS, but are willing to find out. thanks Manrique Matthew Fredrickson escribió: On Jul 25, 2006, at 12:53 PM, Manrique Feoli wrote: Hi all, Here is the situation: A call comes in to an Alcatel PBX and it sends it to an E1 on * , this * either sends the call to a VoIP extension or needs to forward it to an extension back on the Alcatel, but WITHOUT using another slot of the E1 (no tromboning or hairpinning). I've read you can do this with 2b channel transfers implemented on 5ESS, and also on QSIG. I know Matthew Fredrickson did it on * (I think he programmed it for *) I also know there is quite a bit of people pursuing this same goal, which is way important to lower the income barriers for * to enter the legacy world. Has anyone actually done it? I appreciate any input whatsoever, and if possible a sample of how to manage it on *.What to put on the extensions.conf to perform the transfer and any other files needed, Unfortunately, I have not implemented the Q.SIG version of 2b channel transfer, so for the time being you'll have to stick to hairpinning the legs of the call. The Q.SIG version is a little bit more complicated than some of the other versions. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to send a signal via E1/T1 ISDN to asterisk, to ask the call to be moved.
Hi, all I have an * which receives calls from PSTN and some of them fo to an E1 where another system is working (Dialogic Boards). I need to be able to send a signal to * from the system with the Dialogic boards, preferrably via the E1 so that * knows it has to move the call from slot ZAP25 to SIP/ xxx . Im thinking to use the manager with a socket connection for this, but would be much cleaner for me if I can send a message via the E1, has anyone done something similar? please welcome any ideas for this PS (if this sounds familiar, it's because I'm trying to go arround the 2b-channel limitation that we discussed earlier on, where I couldn't find a way to tell ) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
