[Asterisk-Users] fax detect DID variable
Has anyone been able to get the original DID number when a fax comes in and is detected using the Zap fax detect? I am trying to email faxes based on the DID number it came in on but the CALLERID4 variable only contains 's' and not the original number. Someone told be the original id is saved before calling the fax extension. It looks like it is supposed to be in the FAXEXTEN variable but I think the chan_zap.c file is detecting the fax tone and calling the fax extension before it gets saved in the variable. I am using cvs head v1.2.0. I have not updated to the lastest yet because it has been working and it is our office PBX. - James Armstrong ___ --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] fax detect DID variable
Ok, I got the number called in DNID, but now I need to pass it to the
IAXModem so Hylafax can use it to route the email. The use CALLID1-4.
It looks like CALLID 1 2 are the caller and CALLID3 is the
Extension at the time the call is made to the IAXModem extension,
which is 's'. What variable would I set to pass it to CALLID3?
Thanks,
James Armstrong
On Jan 26, 2006, at 3:02 PM, Lee Howard wrote:
James Armstrong wrote:
Has anyone been able to get the original DID number when a fax
comes in and is detected using the Zap fax detect?
In your incoming call context where fax calls come in you need to
Set some user variable to ${EXTEN} before the dialplan continues,
because once the fax is detected it will get changed.
Then, in your Dial (to IAXmodem or whatever) use the user variable
that was used with Set instead of ${EXTEN}.
Lee.
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[Asterisk-Users] feature transfer on PRI
I am having a problem with the tranfer function since changing our t1 to PRI. I used to be able to answer a forwarded call to my cell phone, then transfer the call back using ## (we have it set to ## in features.conf). Since the update it does not work most of the time. At lunch today I had a call and could not get it to transfer. The logs show asterisk getting the DTMF #, then a bunch of stuff about a zap channel not being answered yet. I see two #'s coming in withing the time configured in features.conf but being ignored. It looks like the original call came in on Zap2-1 and asterisk called my cell phone on Zap3-1 and bridged the calls but still thinks Zap2-1 is still dialing and ignores my dtmf even though I am on Zap3-1. Here are a few lines from the log: Jan 25 11:29:20 DEBUG[1131] chan_zap.c: DTMF digit: # on Zap/3-1 Jan 25 11:29:20 DEBUG[1131] chan_zap.c: Dropping frame since I'm still dialing on Zap/2-1... Jan 25 11:29:20 DEBUG[1131] chan_zap.c: Dropping frame since I'm still dialing on Zap/2-1... Jan 25 11:29:20 DEBUG[1131] chan_zap.c: Dropping frame since I'm still dialing on Zap/2-1... Jan 25 11:29:20 DEBUG[1131] chan_zap.c: Dropping frame since I'm still dialing on Zap/2-1... Jan 25 11:29:20 DEBUG[1131] chan_zap.c: Dropping frame since I'm still dialing on Zap/2-1... Jan 25 11:29:20 DEBUG[1131] chan_zap.c: Dropping frame since I'm still dialing on Zap/2-1... Jan 25 11:29:20 DEBUG[1131] chan_zap.c: Dropping frame since I'm still dialing on Zap/2-1... Jan 25 11:29:21 DEBUG[1131] chan_zap.c: Dropping frame since I'm still dialing on Zap/2-1... Jan 25 11:29:21 DEBUG[1131] chan_zap.c: Dropping frame since I'm still dialing on Zap/2-1... Jan 25 11:29:21 DEBUG[1131] chan_zap.c: Exception on 13, channel 2 Jan 25 11:29:21 DEBUG[1131] chan_zap.c: Got event Dial Complete(9) on channel 2(index 0) Jan 25 11:29:21 DEBUG[1131] chan_zap.c: Echo cancellation already on Jan 25 11:29:24 DEBUG[1131] chan_zap.c: DTMF digit: # on Zap/3-1 Anything I can check? This is CVS head as of end of last year. Thanks, James Armstrong ___ --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] Voice mail messages aren't sent to e-mail
I added: mailcmd=/usr/bin/sendmail -f hostname -t to the voicemail.conf file under [general] - James On Jan 6, 2006, at 10:31 PM, Pisac wrote: Yes, I found that this is problem with my server. Second server is connected through second provider, and first server and my domain is hosted at fist provider. My (first) provider has some stupid logic that reject e-mails from mailservers which don't have public hostname but private (my second server has server.local), but accepting all e- mails from it's IP address space (first server). So, my temporary solution was that I set up fake (but existing) hostname for second server (gmail.com), and now my (first) provider accepting e-mails. Very stupid. How you changed mailcmd to add a -f? Did you used nail/mail instead of sendmail, in voicemail.conf? Or maybe some .c source changing? Thanks Pisac I had a similar problem, but I was able to see the message getting rejected to rr.com because they were looking up the hostname pbx and rejecting it. I changed the mailcmd to add a -f realhostname.com and it started working. - James ___ --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] Voice mail messages aren't sent to e-mail
I had a similar problem, but I was able to see the message getting rejected to rr.com because they were looking up the hostname pbx and rejecting it. I changed the mailcmd to add a -f realhostname.com and it started working. - James On Jan 6, 2006, at 4:24 PM, Pisac wrote: Voice-mail messages aren't sent to e-mail address. I have two Asterisk servers, first one is upgraded from 1.0.RC2 to 1.0.9, and second one is from 1.0.7 to 1.0.9. Both Asterisk have EXACTLY same voicemail.conf configuration, but second Asterisk don't sending voice mail messages through e-mail! I'm using almost default voicemail.conf with just one mailbox addedd: 1234 = 1234,MyName,[EMAIL PROTECTED] Why second Asterisk don't sending e-mails? I tested nail program, and I can send any mail without problems. How Asterisk sending mail, through some other program (nail, mail) or by itself? ___ --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] Unable to find key
I think this is normal if you don't have call-forward-busy enabled. They key is deleted when it is disabled and added when enabled. - James Alejandro Vargas wrote: Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' [...] -- Alejandro Vargas ___ --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] HOWOT transfer call from mobile back to extension?
I use this at work. You have to make sure you use the right T / t options when dialing the mobile, then just use the standard # transfer. I changed ours to ##. - James [EMAIL PROTECTED] wrote: Cheers all. I hope I did not miss this in my quick searching for this, and if I did, apologies, please just a minor scolding as you point me at the URL. But, if I did not miss it, then is there any one out there who has figured-out how to skin this cat. I have a few people's mobile phone numbers in call queues, or in follow-me type set-up's, when we want to transfer that call from the mobile phone back to an extension at the office, how best to do that? I happen to (today) be on a Treo650 on the carrier referred to as Stinkular, if it matters. I do not think I can create a three-way call from the mobile, one leg to original caller, and one leg of new call to PBX then enter extension, then get them on phone, and hang-up, since that will drop both legs. Do I need to get Stinkular to add Centrex to my mobile? :) Any all ideas/suggestions/tips/tricks of any kind are very appreciated. (if this grows to a large enough list of tips/tricks, I will distill post to wiki for us all) Thanks very much, Sjobeck www.voip-info.org/tiki-index.php?page=UserPagesjobeck ___ --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] extension seen as busy when it is not
Every few days our receptionist's phone will not take calls on one of the extensions. We have an extension 118 going to the first two lines of her phone and extension 101 going to the other. If we try to dial 118 it goes to voicemail even though she is not on the phone. Asterisk is thinking she is not logged on or something because the message in the log stays there is congestions calling that extension: dialparties.agi: extnum: 118 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: Extension 118 has call waiting enabled2 dialparties.agi: get_dial_string: extnum=[118] -- dialparties.agi: get dial string 118, SIP/118 -- dialparties.agi: DbSet CALLTRACE/118 to 101 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(SIP/101-dc56, SIP/118|25|tTwWr) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing GotoIf(SIP/101-dc56, 0?s-CHANUNAVAIL|1) in new stack -- Executing GotoIf(SIP/101-dc56, 0?s-CHANUNAVAIL|1) in new stack -- Executing NoOp(SIP/101-dc56, Sending to Voicemail box 118) in new stack What can I look at to see why this is happening? Thanks, James ___ --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] Tellabs manuals
I have a 253A manual out of the big three manuals I have, but not the echo canceller. - James C F wrote: Does anybody have a Tellabs manual for: * 253c shelf. the complete model number is: 81.0253c * 2572 Echo Canceller card, complete model number is: 81.2572 I know the wiki has got lots of info on it, but I'm trying to get the original docs from Tellabs. 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
Re: [Asterisk-Users] How do I remove the temp greeting?!?!
0-4-2 Options, Temporary Greeting, Delete Temporary Greeting Matt wrote: Hi, In Asterisk voicemail... if I record a temporary greeting, how the heck do I delete it and go back to using the normal greetings again?! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Bounty Pool
Cool application, one question, can you call from any phone? Is the number on the reference card an example, customer support number, or the number you call into to login? Does the system only call you or is there a way to call in and check the status? Thanks, James Chris Tooley wrote: gNumber has written an application, UnWired Buyer, based on Asterisk. To show our thanks, we would like to extend an offer to the community. We are currently offering an PayPal credit of $10 to everyone that signs up and uses the service within the first 30 days. However if you use the promotion code of ASTERISK when you sign up, your $10 can be diverted into a Bounty Pool to pay for features in Asterisk. A committee of Asterisk developers is being assembled to accept projects and determine when they are complete. UnWired Buyer, our product based on Asterisk, is available at www.unwiredbuyer.com. It is an IVR that allows you to bid on auctions on eBay over a phone call. To earn the $10 for the Asterisk Bounty Pool you need to signup for UnWired Buyer and put ASTERISK in the Promotion Code field. Then within 30 days bid on some auction using UnWired Buyer. You are not required to win the auction but a bid must be placed using the system. For information on the program go to http://www.unwiredbuyer.com/asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] question about priorities?
I'm not sure if Priority is the correct term, but it is the order number as in exten = fax,1- If I have an application that loads / includes another file, will a line of the same order in the included file override the one in the main application? What I need to do is: [test] include custom-test exten = s,1,Playback( exten = 2,2,Hangup [custom-text] exten = s,1,Background(... And have the call come in to s,1 but use the custom code then fall back to s,2 in the main app. I want to modify the way AMP does the Menus / Automated attendant and change one line. It needs to be in the custom file because AMP will overwrite any changes I make to the _additional.conf files when I reload or change anything with AMP. I hope that makes sense, James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer/take call to/from other phone
This is what I use. You pre-pend a '4' to the extension number (I used
that because that is how our old pbx worked). There is a number you can
use that will pickup any ringing extension but I forgot what that is. It
should be listed on the asterisk wiki for Pickup.
exten = _4XXX,1,Pickup(${EXTEN:1})
exten = _4XXX,1,Hangup
- James
Denny Schierz wrote:
hi,
Quoting Chuck Bunn [EMAIL PROTECTED]:
Push the '#' key followed by the extension for a blind transfer.
absolut perfect, thanks :-) .Is there also a shortcut, to take a phone
call from
other phones to me?
cu denny
This message was sent using IMP, the Internet Messaging Program.
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Re: [Asterisk-Users] Queue and agent transfer
This must be similar to a problem I have seen here. Some times the main operator's phone will stop ringing when a call comes in on the queue while the other phones still ring. I have to reset her phone which causes a re-login to get it working again. It must stop after she does an attended transfer. The normal blind transfers are what she does mostly. - James Tamas wrote: Hello, I have a queue for incoming calls with some agents (defined as Agents) using iax2 softphone. I would like to use the Attended transfer (ATXFER) feature, however app_queue cannot handle it (I guess because it is not a channel). For this reason I put a Local channel in between with /n option. This way ATXFER works perfectly. The problem is that when the call has been transferred, the originally dialed agent does not get freed up - this is from some point of view logical. So the queue sees the agent as engaged/busy, while he/she is not anymore. Does anybody have any idea how to free up the not used agent? (transferer) Thank you in advance! Kind regards, Tamas ___ --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] Small office with all employee's offsite
Jason Marshall wrote: OK, then this is easy. Instal Asterisk in the central location, along with a Sipura SPA-3000. Configure that unit to answer the incoming POTS line and act as a VOIP gateway for Asterisk. Then configure two additional SPA-3000 units, one at each employee's location. Then, configure Asterisk (I recommend [EMAIL PROTECTED] for your setup, BTW) to route the incoming call to the right extension based on time of day, auto-attendant, whatever. The SPA-3000 units at each remote site will also be able to accept the employee's incoming POTS line and pass that call through to the phone they normally use without resorting to sending it to the Asterisk server and back. (It's all in the SPA-3000 setup. Very cool indeed. Thanks Tom! Now to throw a monkey-wrench into the works... One of the employees spends a lot of time outside of his home office, and is then reachable only by cell phone. But we (for obvious reasons) don't want to hand out his cell number to everyone who wants to reach him. So, he will often forward his home phone to his cell, and forward the main office number to his home number (so when people call the office, they get his cell without realizing it). We do this all the time. We just moved and have three people working from their homes. The boss's extension rings here locally on a spare phone and rings his IAX2 phone at home. He also forwards his extension to his cellphone when he is out using *72 on the Asterisk box. One employee is working from out of state and his extension calls his cellphone. When someone dials his DID number it dials back out to his cell phone and no one knows any different. When we dial his three digit extension here it goes to his cell phone. The last person has an IAX client running on his laptop and takes calls from there. When someone calls in and presses '2' for support it rings a guy out in production and the other person working from home. I have my extension set to ring my Grandstream phone and my cell phone at the same time and I can take the calls from anywhere. I can even transfer a call back to another extension from my cellphone if they need someone else. Asterisk does all the call forwarding and phone routing. - James Is there any way to use the SPA-3000 at his house to re-route calls (VOIP calls, in this case) to his cell? Or would that have to be done at the office where the server is physically. I'm not clear on whether the Asterisk server can control a remote SPA-3000 in this way. As long as Asterisk has a way to re-dial out a phone line or voip provider, it can route an extension anywhere and the caller will not know it. I guess this could be done directly from the Asterisk server, couldn't it? It wouldn't be something that could happen automatically; it would have to be manually turned on and off. But it would also require another POTS line at the main office for the outbound call -- so I'd rather leverage the phone line at his home office to make the outgoing call to his cell phone if at all possible... One more monkey-wrench -- what if I want both of the employees to be on the phone at the same time? Two incoming POTS lines, and two SPA-3000's at the office? Or does it make more sense at that time to get a TDMxx card? This will not change, you're still looking at three lines in the scenario I outlined above. (Unless you switch to incoming VOIP, but I do *NOT* recommend that.) Nope, I don't believe in VOIP replacing POTS completely yet. Maybe in 5 years... =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | Jason Marshall, [EMAIL PROTECTED] Spots InterConnect, Inc. Calgary, AB | =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- ___ --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] FAX difference IAXModem / Hylafax and spandsp app_rxfax
I looked into why I can't get the original DID number called when a fax
is detected (so I can later route to the correct email address). There
is a variable called FAXEXTEN that is created when a fax is detected,
but it is not being populated with the original extension / did number
called. It always has 's' as the original extension. Does anyone know
how to fix this? I just want to use the zap fax detection so each person
can have their personal DID number accept faxes and calls, then route
the fax to their email address based on the DID number called.
- James
James Armstrong wrote:
In this example faxdetect is overwriting the DID.
So the trick then would be to somewhere early-on in your dialplan grab
the DID into some variable, and then restore it after the fax
detection occurs...
[default]
exten = _X.,1, SetVar(ORIGEXTEN=${EXTEN})
exten = s,2,Wait(3)
.
exten = fax,1,Dial(IAX2/ttyIAX0/${ORIGEXTEN})
Lee.
I am having no luck here. It seems the fax detection is overriding
everything. I added the above and it never gets called. I guess it might
be time to look at the Asterisk code and see if I can create another
variable before the redirect happens.
-- Starting simple switch on 'Zap/1-1'
-- Redirecting Zap/1-1 to fax extension
-- Executing Dial(Zap/1-1, IAX2/999/|20) in new stack
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Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax
Found out why there is no original DID set. It looks like while waiting
for the incoming digits timeout (DID), we are getting the fax tone
detect and it is sending a digit 'f' which immediately starts the fax
extension before the incoming DID has been saved.
Is there a way to set in the zap config how many digits we are receiving
so there is not timeout waiting for the last digit? We only get 7 digits
coming in.
- James
James Armstrong wrote:
I looked into why I can't get the original DID number called when a fax
is detected (so I can later route to the correct email address). There
is a variable called FAXEXTEN that is created when a fax is detected,
but it is not being populated with the original extension / did number
called. It always has 's' as the original extension. Does anyone know
how to fix this? I just want to use the zap fax detection so each person
can have their personal DID number accept faxes and calls, then route
the fax to their email address based on the DID number called.
- James
James Armstrong wrote:
In this example faxdetect is overwriting the DID.
So the trick then would be to somewhere early-on in your dialplan
grab the DID into some variable, and then restore it after the fax
detection occurs...
[default]
exten = _X.,1, SetVar(ORIGEXTEN=${EXTEN})
exten = s,2,Wait(3)
.
exten = fax,1,Dial(IAX2/ttyIAX0/${ORIGEXTEN})
Lee.
I am having no luck here. It seems the fax detection is overriding
everything. I added the above and it never gets called. I guess it
might be time to look at the Asterisk code and see if I can create
another variable before the redirect happens.
-- Starting simple switch on 'Zap/1-1'
-- Redirecting Zap/1-1 to fax extension
-- Executing Dial(Zap/1-1, IAX2/999/|20) in new stack
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[Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax
I was wondering if anyone would know why receiving a fax via an extension - IAXModem - Hylafax would work, but fax - rxfax (spandsp) does not. When I try to use the app_rxfax the fax machine reports a comm error. If I use the IAXModem client connected to Hylafax it works. I don't mind the Hylafax method, but I want to be able to route the fax/pdf to a mail address based on the extension/DID called. As it is right now, Hylafax does not know which DID was called and only sends notifications to a general mailbox. cvs 1.2 Thanks, James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax
Thanks, that helps. What variable (in Asterisk) would I modify if I wanted to tweak one of the CALLERID fields that gets passed? Right now, it is not passing the DID number the fax came in on, it is always passing 's' as the NDID. - James Lee Howard wrote: James Armstrong wrote: I was wondering if anyone would know why receiving a fax via an extension - IAXModem - Hylafax would work, but fax - rxfax (spandsp) does not. When I try to use the app_rxfax the fax machine reports a comm error. If I use the IAXModem client connected to Hylafax it works. HylaFAX supports ECM in Class 1, and so undoubtedly the fax session with rxfax was quite different from the fax session with HylaFAX. So it's not a perfect apples-to-apples comparison, but I would generally consider HylaFAX to be more mature than rxfax/txfax. The fact that the fax works with iaxmodem but not with rxfax means that the error is not with spandsp (the library), but rather with the rxfax application. I don't mind the Hylafax method, but I want to be able to route the fax/pdf to a mail address based on the extension/DID called. As it is right now, Hylafax does not know which DID was called and only sends notifications to a general mailbox. Then you're missing something. HylaFAX does (or at least could) know the extension that was dialed. You should be using HylaFAX 4.2.2 or 4.2.3, and your modem config file should have these lines in it: ModemResetCmds: AT+VCID=1 CallIDPattern: NDID= That will parse the modem output for NDID= responses and put them into the CallID array which is passed to faxrcvd and FaxDispatch. So if you're using the default modem config file provided with IAXmodem then you'd refer to $CALLID4. See 'man callid' for a more detailed explanation. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax
Thanks, that helps. What variable (in Asterisk) would I modify if I wanted to tweak one of the CALLERID fields that gets passed? Right now, it is not passing the DID number the fax came in on, it is always passing 's' as the NDID. I think the key is to just not alter the DID to begin with: exten = 5551212,1,Dial(IAX2/ttyIAX0) The only problem here is, we are using our normal did numbers that forward to our extensions. Asterisk detects the fax tone and sends it to the exten = fax,1, extension. instead of: exten = s,1,Dial(IAX2/ttyIAX0) I will try setting the caller id based on the DID number that was called and see if that works. If you need some kind of pattern matching, then use it, but using s rewrites the DID. Thanks, James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax
The TIFF - PDF conversion is working fine, just the mime-encoding/attachment stuff to the email seems to be where i am having issues. I am able to use the existing conversion stuff from the command-line. I have metamail, uuencode, and base64-encode installed, but must be missing something :) ___ I fought this problem yesterday. I did not have base64-encoding or mimencode. Once I put those on it still did not work. It turned out to be the setup.cache file had a ENCODING='', once I changed it to ENCODING='base64' it worked. That is used to generate the Content-Type-Encoding header in the email. - James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax
Thanks, that helps. What variable (in Asterisk) would I modify if I
wanted to tweak one of the CALLERID fields that gets passed? Right
now, it is not passing the DID number the fax came in on, it is
always passing 's' as the NDID.
I think the key is to just not alter the DID to begin with:
exten = 5551212,1,Dial(IAX2/ttyIAX0)
The only problem here is, we are using our normal did numbers that
forward to our extensions. Asterisk detects the fax tone and sends it to
the exten = fax,1, extension.
instead of:
exten = s,1,Dial(IAX2/ttyIAX0)
I will try setting the caller id based on the DID number that was called
and see if that works.
If you need some kind of pattern matching, then use it, but using s
rewrites the DID.
Does Asterisk have a variable that keeps the original ${EXTEN} anywhere?
I tried to use the exten = fax,1,Set(CALLERID(number)=${EXTEN})
function to set the original DID number that the call came in on, but by
the time it gets to the fax, the Extension is already set to 'fax' and I
do not know the DID number that was called. I also tried to use
Set(CALLERID(number)=${FROM_DID}), but it looks like Asterisk is
detecting the fax tone and diverting the call before it even gets to the
[from-did] section of my extensions_additional.conf file.
Thanks,
James
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Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax
In this example faxdetect is overwriting the DID.
So the trick then would be to somewhere early-on in your dialplan grab
the DID into some variable, and then restore it after the fax detection
occurs...
[default]
exten = _X.,1, SetVar(ORIGEXTEN=${EXTEN})
exten = s,2,Wait(3)
.
exten = fax,1,Dial(IAX2/ttyIAX0/${ORIGEXTEN})
Lee.
I am having no luck here. It seems the fax detection is overriding
everything. I added the above and it never gets called. I guess it might
be time to look at the Asterisk code and see if I can create another
variable before the redirect happens.
-- Starting simple switch on 'Zap/1-1'
-- Redirecting Zap/1-1 to fax extension
-- Executing Dial(Zap/1-1, IAX2/999/|20) in new stack
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Re: [Asterisk-Users] Needed - Pager notification script
I have one also that just does nag paging. It looks up the extension in the db and gets the pagers to notify. Sets x number of attempts and if a user checks his messages it will clear the remainder of the pager attempts. Written in perl. Not a daemon, uses the run_external_notify. Sends one message immediately with the message and caller id info, the nag pages are sent with just 'Mailbox xxx has y messages'. I would be interested in looking at the daemon version. - James B. J. Bomar wrote: I have a script that doesn't quite fit your needs, but does send out email reminders for on a regular basis, and runs as a daemon. If you are interested, please let me know and I will send it to you. A little warning, this was one of my first major perl scripts, so it may be a little ugly and crude. :) B. J. -Original Message- From: Tom Rymes [mailto:[EMAIL PROTECTED] Sent: Thursday, November 10, 2005 16:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Needed - Pager notification script This is a second post on this subject from me: A while back, someone posted to the list about a script that they had created that would handle paging and escalation for on-call mailboxes. Basically, it would monitor the voicemail directories and if a message was left and not retrieved by the on-call tech within X minutes, the system would page the tech again. If after Y minutes the message had still not been retrieved, the script would then page his/ her supervisor, and so on. Unfortunately, the original poster did not include the script body in his post to the list and it is not available at the wiki. My question: Does anyone have such a script that they have already created? Would you be willing to share? If not, what if I chipped in some $$$. I'm really trying to avoid reinventing the wheel here Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues with one Agent set to DND
I have a question. Is there any way to have a caller entering a Queue to go to voicemail if there is only one Agent and that extension has the phone set to DND? We have one extension that is the primary service technician and have it set to always be a member / logged in, so he cannot just logout when he goes to lunch. The phone rings when he is at lunch and drives people crazy. I tried setting DND on, when a call comes into the queue it shows his extension as do not disturb and sets it to BUSY, but the call is still on hold. I would think that if there is only one agent and that agent is set to DND the call should proceed as if there were no agents logged in. - James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues with one Agent set to DND
Tried that. The queue has a static agent of SIP/107. When calling the queue it shows 107 as being BUSY (DND enabled). The caller just stays in the queue. What I really need is to have the caller stay in queue when the extension is busy (because that is that queues are all about), but have the caller leave the queue if DND is enabled in the database for that extension and there is only one extension as an Agent. - James Juan Manuel Coronado Z. wrote: James Armstrong wrote: I have a question. Is there any way to have a caller entering a Queue to go to voicemail if there is only one Agent and that extension has the phone set to DND? We have one extension that is the primary service technician and have it set to always be a member / logged in, so he cannot just logout when he goes to lunch. The phone rings when he is at lunch and drives people crazy. I tried setting DND on, when a call comes into the queue it shows his extension as do not disturb and sets it to BUSY, but the call is still on hold. I would think that if there is only one agent and that agent is set to DND the call should proceed as if there were no agents logged in. - James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users James, You should try to use normal sip channels instead of agents and define them as members of a queue, so you are able to set a voicemail when busy/unavailable. Check http://www.voip-info.org/wiki/view/Asterisk+call+queues * *// Members are those channels that are active answering the Queue. It can be agents or normal channels, like sip/snom23 Regards, Juan Manuel Coronado Z. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New package posted to Sourceforge
Looks good except one problem I am having. The AMP script does not store the info. It adds a blank speeddial. If I edit the database the AMP script will show the correct info, but it never updates the fields. - James Paul wrote: I just posted a few addons for the AMP users ... These are several routines I found necessary for my system 1: Speed Dials revised my way (AMP front end into DB), 2: Intercom in business, 3: Group Paging in business, 4: Cisco phone display (XML) of internal directory list from AMP extensions DB. Intercom Paging is not AMP dependent - tested on Cisco phones. http://sourceforge.net/projects/enhanceme/ Paul Norris Silicon Valley Products ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New package posted to Sourceforge
I have something setup wrong. Everything in AMP works except this script. The Submit button does not do anything. Here are some errors I am getting, I don't know what they mean, I have checked permissions on the session directory and they look ok: [client 10.100.1.202] PHP Warning: session_start(): open(/var/lib/php/session/sess_eaae9fabf2d5749ae5f01ff7c5a7b4fd, O_RDWR) failed: Permission denied (13) in /var/www/html/admin/config.php on line 23, referer: http://10.100.1.225/admin/config.php?display=14 [client 10.100.1.202] PHP Notice: Undefined index: lang in /var/www/html/admin/header.php on line 46, referer: http://10.100.1.225/admin/config.php?display=14 [client 10.100.1.202] PHP Notice: Undefined index: action in /var/www/html/admin/speeddial.php on line 58, referer: http://10.100.1.225/admin/config.php?display=14 [client 10.100.1.202] PHP Notice: Undefined index: clk_reload in /var/www/html/admin/footer.php on line 26, referer: http://10.100.1.225/admin/config.php?display=14 [client 10.100.1.202] PHP Warning: Unknown(): open(/var/lib/php/session/sess_eaae9fabf2d5749ae5f01ff7c5a7b4fd, O_RDWR) failed: Permission denied (13) in Unknown on line 0, referer: http://10.100.1.225/admin/config.php?display=14 [client 10.100.1.202] PHP Warning: Unknown(): Failed to write session data (files). Please verify that the current setting of session.save_path is correct (/var/lib/php/session) in Unknown on line 0, referer: http://10.100.1.225/admin/config.php?display=14 [client 10.100.1.202] PHP Notice: Undefined index: lang in /var/www/html/admin/common/script.js.php on line 7, referer: http://10.100.1.225/admin/config.php?display=14 - James Paul wrote: After you place one in you MUST submit. That is only when it is saved Paul -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Armstrong Sent: Tuesday, November 08, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] New package posted to Sourceforge Looks good except one problem I am having. The AMP script does not store the info. It adds a blank speeddial. If I edit the database the AMP script will show the correct info, but it never updates the fields. - James Paul wrote: I just posted a few addons for the AMP users ... These are several routines I found necessary for my system 1: Speed Dials revised my way (AMP front end into DB), 2: Intercom in business, 3: Group Paging in business, 4: Cisco phone display (XML) of internal directory list from AMP extensions DB. Intercom Paging is not AMP dependent - tested on Cisco phones. http://sourceforge.net/projects/enhanceme/ Paul Norris Silicon Valley Products ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New package posted to Sourceforge
I think my problem is Submit is not doing anything. I fixed all errors except the Unknown index errors in the httpd log. When I click Submit nothing happens. If I comment the tests out of the checkSpeednr() and just have the .submit it works ok. - James Paul wrote: Just tested it again. That happens when you fill in the first line then hit add before submit. Please use the submit key first then add a new item. :) Paul -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Armstrong Sent: Tuesday, November 08, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] New package posted to Sourceforge Looks good except one problem I am having. The AMP script does not store the info. It adds a blank speeddial. If I edit the database the AMP script will show the correct info, but it never updates the fields. - James Paul wrote: I just posted a few addons for the AMP users ... These are several routines I found necessary for my system 1: Speed Dials revised my way (AMP front end into DB), 2: Intercom in business, 3: Group Paging in business, 4: Cisco phone display (XML) of internal directory list from AMP extensions DB. Intercom Paging is not AMP dependent - tested on Cisco phones. http://sourceforge.net/projects/enhanceme/ Paul Norris Silicon Valley Products ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail issue with beta1 beta2
I have a similar issue. Using beta, small group of users. It has happened to me a few times. When I go to play the new messages, it gives the header, then plays the menu. If I press 5 to relisten to the message it plays correctly. - James On Nov 8, 2005, at 9:12 PM, [EMAIL PROTECTED] wrote: I'm having a voicemail problem that I can't seem to eradicate, and I'm hoping to get some direction from this list. The phone system in question is using 1.2 Beta 2, using Realtime with about 5k users in the database. There have been a few messages that appeared to be blank, but when they forwarded to another extension with a prepended message, the original message would appear. Apparently some voicemail messages have an empty msg,txt file, which the voicemail system recognizes as a voicemail, but the corresponding msg.wav file won't play. When the user forwards what they believe is a blank message to my mailbox, the msg.txt file is rewritten, and I receive the prepended message followed by the original message. I didn't see this kind of issue in the archives, or in google. Please advise Thanks! Niles ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP and voicemail passwords
Anyone here using AMP and having problems with users chaning their voicemail passwords? They stick until I go into AMP and make changes then reload. The AMP settings contain the old password and are overwriting the new one saved by the user. What am I doing wrong or what is the correct way to do it? - James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time based call direction
I have a few lines of code that check the current date (20051103) for a
database entry in HOLIDAYS and if it is there go to the 'Night'
attendant for business closures on holidays.
exten = s,4,SubString(TODAY=${DATETIME},0,8)
exten = s,5,DBGet(INHOLIDAY=HOLIDAYS/${TODAY})
exten = s,6,GotoIf($[${INHOLIDAY}=1]?from-pstn-afthours,s,1:)
- James
Kyle Hagan wrote:
Adam Moffett wrote:
include = atlunchcontext|11:00-11:59|mon-fri|*
include = notatlunchcontext|09:00-10:59|mon-fri|*
include = notatlunchcontext|12:00-18:00|mon-fri|*
include = afterhourscontext|18:01--8:59|mon-fri|*
I wasn't aware that include allowed a time qualifier. Does that mean
that the specified context will only be included at the specified time?
Correct. We have been useing that here for some time now.
Kyle
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Re: [Asterisk-Users] iax softphone
How about any IAX softphones for the pocket pc platform? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
