Re: [asterisk-users] 9 becomes 99 ? And other strangeness
Turn off relaxdtmf in zapata.conf if that does not help play with the
rxgain, if that does not help, play with the txgain. If the volume is
too loud or too soft on zap channels, Asterisk can sometimes miss or see
double DTMF.
Brian Candler wrote:
On Thu, Sep 14, 2006 at 10:37:59AM -0500, Rich Adamson wrote:
Brian Candler wrote:
On Thu, Sep 14, 2006 at 09:00:57AM -0500, Rich Adamson wrote:
[outbound]
exten = _9.,1,Dial(Zap/4/${EXTEN:1}) NOTE HERE
exten = _9.,2,Congestion()
exten = _9.,102,Congestion()
Try replacing the first step above with:
exten = _9.,1,Dial(Zap/4/w${EXTEN:1})
Note the w in the above means wait for about a 1/4 second before
sending the number to the central office.
Some central offices are not ready to receive digits as quickly as
asterisk sends them out.
Interesting feature, thank you, but I don't think that's the problem.
Notice that Asterisk's own log shows that it thinks the number called is
99X and therefore dials out to 9X, where in fact I only dialled
9X and so it should be dialling X.
Console:
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1]
Dial(Zap/1-1,Zap/4/907974XX) in new stack
-- Called 4/907974XX
-- Zap/4-1 answered Zap/1-1
-- Native bridging Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
== Spawn extension (internal, 9907974XX, 1) exited non-zero on
'Zap/1-1'
-- Hungup 'Zap/1-1'
If this were consistent I could use ${EXTEN:2} to strip off the two 9's,
but
it isn't.
Try the above an see what the result is. If it does not address the
problem, at least one item has been removed from the list of
possibilities. ;)
Without wishing to be contrary: could you explain to me how it could
possibly work?
Asterisk thinks I dialled (on the inbound leg) a number starting 99. After
it has got this wrong number, it then dials it on the outbound leg,
stripping off one of the 9's as that's what the dialplan says.
Adding a 1/4 second delay on the outbound leg won't change the fact that
it's trying to dial the wrong number in the first place.
Regards,
Brian.
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Re: [asterisk-users] Tracking the source of a disconnect?
Doug Lytle wrote: Jamin W. Collins wrote: Doug Lytle wrote: callprogress = yes The only thing I'm iffy about is the above entry. Maybe it's mistaking the progress as disconnect? You should never, ever use callprogress or busydetect when using a PRI. In fact, you could not use it in general as both options can cause random disconnects. ___ --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] Tracking the source of a disconnect?
Jamin W. Collins wrote: Doug Lytle wrote: Jamin W. Collins wrote: Doug Lytle wrote: callprogress = yes The only thing I'm iffy about is the above entry. Maybe it's mistaking the progress as disconnect? The calls in question are connected for varying time frames. In some cases 5 minutes, some 20 seconds, others 40 minutes before the disconnect occurs. This still a concern? Yes. Don't use callprogress or busydetect. PRIs have both as out of band features. ___ --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 / Patton SmartNode SN2400 Strangeness FYI
You should never have callerid=xxyyzz as some devices (as you just discovered) choke on the since that is not a valid Caller*ID character. I think some versions of the Cisco phone SIP firmware also has a similar problem. George Pajari wrote: Just a short problem description/resolution so others do not have to experience the same frustration and lost hours. Set Up: Sipura devices (various, mostly SPA2100, SPA2102) connecting to an Asterisk box (1.2.4). Asterisk box talks to a Patton SmartNode SN2400 SIP/PRI gateway. Inbound calls work fine. Outbound calls work fine from some units, not others. Problem: if sip.conf contains callerid=xxyyzz it works; if it contains callerid=xxyyzz then the SN2400 accepts the SIP INVITE but refuses to handle it. Do not have the patience, time, or inclination to run more SIP debugs to see exactly what is happening -- just posting this in the hope it might help someone somewhere. ___ --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] problems with Polycom 500 boot up
Forum wrote: I have a Polycom 500 that I am having issues with provisioning via an ftp server. I have a bunch of 301’s that find the server and configure without an issue. For some reason the 500 gives me an error that it ‘could not contact boot server’ and will reboot continuously. I also get the error ‘Error updating Bootrom’. I am using Bootrom 3.2.1. What files do I need on the ftp server ? – I have sip.Id, bootrom.Id, sip.ver, phone1.cfg and sip.cfg. For some weird reason many of our new Polycom phones use 456 as their default FTP password, instead of what the standard password usually is. We have not eliminated human intervention as the cause of this, but I don't think this is the case. ___ --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] callback without agi
You can't dial from exten = h
You could use an AGI with a .call file, or you could create the .call
file from inside the Asterisk dialplan. Heck, you could do it with
System() commands. See sample.call in the asterisk source directory. as
well as docs/ in the asterisk source directory.
Patricio Valarezo wrote:
Hi, it's possible to implement a callback without agi?, i'm trying this
but * exits without dialing (if I hungup during s,3 wait) but if it
hungs in s,4 it dials, so is there an explanation to this behavior?
there is an alternative to do it? just for learning
thanks for your answers
[followme]
exten = s,1,NoOp(Followme me sigue)
exten = s,2,NoOp(El CID es ${CALLERID(num)})
exten = s,3,Wait(4)
exten = s,4,Hangup()
; al cortar debera iniciar la secuencia
exten = h,1,NoOp(${CALLERID(num)} ha cortado)
exten = h,n,NoOp(channel es ${CHANNEL})
exten = h,n,Wait(10)
exten = h,n,NoOp(aqui podriamos marcar)
exten = h,n,Dial(${CANAL}/${CALLERID(num)})
exten = s,n,Hangup()
PV.
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Re: [asterisk-users] University switches to Asterisk
What other ones are there? Porier, Jeremy M. wrote: They're not the only ones :-) Jeremy Porier Senior Director of Information Systems and Technology Colorado Christian University [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, September 13, 2006 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] University switches to Asterisk Interesting article I found linked from Groklaw: Sam Houston State University replaces Cisco CallManagers, Nortel PBXs with Linux-based VoIP and messaging servers http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1 Doug ___ --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] All circuits are busy now???
BerkHolz, Steven wrote: All circuits are busy now makes perfect sense in my PRI trunk is full. How do I stop asterisk from playing this recording when it is a wrong/bad number? I gat a call today that a user was trying all day to call a number in Mexico and kept getting the above recording. I said, try in on your cell phone, and they received a this number is not is service. I would like to either hear the far recording (I think I will get billed for this), or internally play a different message. I think the issue is that I am using a PRI and am receive the cause code that is triggering the above recording. Can asterisk play a different message for this? and only play the above message if MY circuit is busy? Sounds like you are using some Asterisk GUI. Can't help with that and this message will only be useful to others. When Dial exits it will set the value of HANGUPCAUSE to something. Use the dialplan to play different messages depending on the value of hangupcause. See show application dial in the Asterisk CLI, /path/to/src/asterisk/docs/README.variables, http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf for the cause code values, and /path/to/src/asterisk/include/asterisk/causes.h for which causes Asterisk knows about. ___ --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] What don't I get about SIP?
Mike wrote: Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Because the i extension is for IVRs and things like that. ___ --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] No dialtone, just directly busy
Remove immediate=yes from /etc/asterisk/zapata.conf Henrik Woffinden wrote: That's exactly what happens: When I pick up the handle, this is what I get: -- Extension 's' in context 'from-inside' from '11' does not exist. Rejecting call on channel 0/2, span 2 Do you know what to do in the dialplan? Best regards, Henrik Woffinden Tim St. Pierre wrote: Could you send us some CLI output? Look for something like this Invalid extension s in context whatever your dial context is It could be that lifting the handset without dialing is opening a channel to the s extension, since there are no digits being dialed. There is a workaround for this, but it means creating a dialplan that produces dialtone and waits for digits. -Tim On September 8, 2006 14:44, Henrik Woffinden wrote: Hi, I'm using Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I've got 3 ISDN phones attached. When I want to dial out I can do it in 2 ways.. 1) Type in number with handle still on.. Lift handle and we dial the number 2) Lift handle and then press the number Both methods should work, but only the first does. With the second I expected a dialtone but it goes immedately to busy signal. No dialtone first. Why is that? ___ --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] No dialtone, just directly busy
For some reason your phone is dialing an empty extension as soon as you go off hook. exten = s would be the same as exten = '' Henrik Woffinden wrote: immediate is already set to immediate=no, so that's not it. Best regards, Henrik Woffinden Eric ManxPower Wieling wrote: Remove immediate=yes from /etc/asterisk/zapata.conf Henrik Woffinden wrote: That's exactly what happens: When I pick up the handle, this is what I get: -- Extension 's' in context 'from-inside' from '11' does not exist. Rejecting call on channel 0/2, span 2 Do you know what to do in the dialplan? Best regards, Henrik Woffinden Tim St. Pierre wrote: Could you send us some CLI output? Look for something like this Invalid extension s in context whatever your dial context is It could be that lifting the handset without dialing is opening a channel to the s extension, since there are no digits being dialed. There is a workaround for this, but it means creating a dialplan that produces dialtone and waits for digits. -Tim On September 8, 2006 14:44, Henrik Woffinden wrote: Hi, I'm using Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I've got 3 ISDN phones attached. When I want to dial out I can do it in 2 ways.. 1) Type in number with handle still on.. Lift handle and we dial the number 2) Lift handle and then press the number Both methods should work, but only the first does. With the second I expected a dialtone but it goes immedately to busy signal. No dialtone first. Why is that? ___ --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 ___ --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] What don't I get about SIP?
Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --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] I'm I wrong - No 3-way calling for Single line sets?
You do not mention the device you are using. I'll assume Zap. Enable three way calling and conference in zapata.conf then use FLASH. Bart Fisher wrote: It appears the only way to cause a 3-way call (or a screened transfer) is by using conference - nasty This mean SLT would need to transfer to conference than add second party, then add themselves. I've searched and I can't find anything that works in asterisk like the Telco method or am I blind? ___ --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] What don't I get about SIP?
Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --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] What don't I get about SIP?
Then you are doing something else wrong. If the call gets to Asterisk then the exten = lines I gave should match if they are in that context. I use this all the time. Mike wrote: But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --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 ___ --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] What don't I get about SIP?
What IS your Polycom dialplan, and do you have the digit.impossiblematch set? Eric ManxPower Wieling wrote: Then you are doing something else wrong. If the call gets to Asterisk then the exten = lines I gave should match if they are in that context. I use this all the time. Mike wrote: But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: No such device - TDM13B
wcfxo is used only for the X100P (and some clones). It is not used for any other card. wctdm supports both FXO and FXS. Maybe you are just confused about which module is associated with which channel. Iván Vega R. wrote: Upon further investigation, I tried the following: lsmod | grep 'wc*' I can see wctdm (I believe this is the fxs module, no?), wcfxo, zaptel... so I think so far so good. Then: cat /proc/interrupts There I only see the wctdm: 50:9556184 0 IO-APIC-level uhci_hcd:usb3, wctdm I reckon I should move the card to another slot or somehow disable the USB port? Also, wcfxo does not show up on the list, and I can see on the card that the FXO port (I'm guessing it's that port) is not lit, while the other 3 are. So I'm guessing it didn't get installed or something? The plot thickens... Does anyone have any idea what the problem could be? Thanks! On 9/8/06, Iván Vega R. [EMAIL PROTECTED] wrote: Hi again! So while ztcfg ran without errors, now asterisk won't run. Here's the relevant part of the error: WARNING[13055]: chan_zap.c:921 zt_open: Unable to specify channel 1: No such device ERROR[13055]: chan_zap.c:6879 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Now I'm totally lost, so any help is really appreciated. Thanks! Ivan V. ___ --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] What don't I get about SIP?
I don't know if this helps, but this is how my 80+ Polycom phones are set up. dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=0 digitmap dialplan.digitmap=9,1[2-9]xx[2-9]xx|9,[2-9]xx|[2-8]xxx|9,[2-9]11|911|1x|9,011x.T|*xx dialplan.digitmap.timeOut=5/ Mike wrote: Here it is: dialplan dialplan.impossibleMatchHandling=1 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=[7]xx|[9]xxT|[9][1]xxT dialplan.digitmap.timeOut=3/ When I dial 845, I get fast busy. When I dial 9-555-555-, it dials without the need to press send. All good result. When I dial 9-555-5 and wait, nothing happens Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 7:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? What IS your Polycom dialplan, and do you have the digit.impossiblematch set? Eric ManxPower Wieling wrote: Then you are doing something else wrong. If the call gets to Asterisk then the exten = lines I gave should match if they are in that context. I use this all the time. Mike wrote: But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --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 ___ --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 ___ --Bandwidth
Re: [asterisk-users] Response to KP Flemming...
Andrew Kohlsmith wrote: On Thursday 07 September 2006 08:45, Matt Riddell (IT) wrote: When and where did KPF admit to it being Digium's code? Via psychic vibrations, obviously. It's not Digium's code, IIRC. It's ITU code. You can download the ITU reference code (in C) from the ITU for free. You can't USE it, because you need a license from the patent holders, but the source code for these is not a big secret. ___ --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] why executed Hangup doesn't exit DialPlan?look my dialplan...
Your problem is caused by using exten = _. DON'T DO THAT!
When Hangup() is being run then Asterisk will jump to exten = h Since
_. will match h it will go there.
Marco Mouta wrote:
Hi all,
I think i'm missing something very very basic! I want my calls with DID
48XX
(From pstn E1 TE110P) to be answered then playback a file and hangup.
Part of my extensions.conf where from-pstn is the context for all calls
from
pstn line is:
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did-custom
include = from-pstn-timecheck ; this has to be included otherwise
it overrides ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)
[ext-did-custom]
exten = _48XX,1,Answer
exten = _48XX,n,SetVar(FROM_DID=${EXTEN})
exten = _48XX,n,Playback(vm-goodbye)
exten = _48XX,n,Hangup
[from-pstn-timecheck]
exten = _.,1,Goto(s,1) ; catch-all matching for calls that have
DID
info (if a DID route hasn't matched them)
exten = s,1,GotoIf($[${IN_OVERRIDE} =
forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} =
forceafthours]?from-pstn-afthours,s,1:)
exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)
exten = s,4,Goto(from-pstn-afthours,s,1)
Problem, look my Asterisk CLI :
-- Accepting call from '2132' to '4888' on channel 0/1, span 1
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SetVar(Zap/1-1, FROM_DID=4888) in new stack
-- Executing Playback(Zap/1-1, vm-goodbye) in new stack
-- Playing 'vm-goodbye' (language 'pt')
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (from-pstn, 4888, 4) exited non-zero on 'Zap/1-1'
-- Executing Goto(Zap/1-1, s|1) in new stack
-- Goto (from-pstn,s,1)
-- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in
new stack
-- Goto (from-pstn-reghours,s,2)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing PlayTones(Zap/1-1, ring) in new stack
-- Executing NVFaxDetect(Zap/1-1, 8) in new stack
-- Channel 0/1, span 1 got hangup request
After the hangup the call seems to keep executing Dialplan why?? Does
this is related with autofallback option in globals???
Why Hangup didn't exit dialplan?
Hope some one can help me.
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Re: [asterisk-users] does anyone offer truly unlimited voip in the US
Steven M. Sawczyn wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm realizing that unlimited really is in fact limited -- Galaxy Voice's unlimited plan, for example, translates to a mere 2500 minutes/month. In researching other SIP providers, I'm finding that their terms of service define unlimited as something similar. Does anyone know of a provider in the US that turly offers unlimited calling, or segnifigantly more than 2500 minutes/month? Most providers have unlimited minutes on the plans that are not flat rate. i.e. you can use as many mins as you want at 2/cents/min. If you mean unlimited for a flat monthly fee there is nobody out there stupid enough to offer that service, or, if they are, they don't stay in business. ___ --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] Max number of SIP devices registered to anextension
This is why we set the SIP user ID to be the MAC of the device. It helps us remember that EXTENSION != DEVICE. Joshua Colp wrote: Brandon Galbraith wrote: I'm attempting to have multiple phones (geographically seperated) register to a single extension, so when the extension is dialed, any phone can pick up the call. Is this better handled by having each phone have a seperate extension, and handle the call routing in a dial plan? -brandon It might be wise to disassociate the term extension from a device... because in Asterisk an extension is a set of instructions that execute a set of applications. For your need you should probably have multiple devices in sip.conf, and have the extension dial all of them. ie: Dial(SIP/145_1SIP/145_2SIP/145_3) 3 phones would each be registered on the machine as 145_1, 145_2, and 145_3. The first one to pick up would get the call and all the rest would stop ringing. ___ --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 with PABX
Dean Collins wrote: Yes it is possible. May I suggest you spend more time with www.voip-info.org Or even better download www.trixbox.org on an old server to get an idea of how configs work. Getting Trixbox would help him understand how Trixbox configs work, not how Asterisk configs work. ___ --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] NAT problems
andrutto wrote: Hi, Does anyone know how to solve this issue. I have Asterisk box on public IP and three clients connected to it. Unfortunately they are behind NAT (simple one-to-one). Those three clients can make outgoing calls hassle free, but when I try to make a call between them something is not right. I am using Linksys PAP-2 (two clients are connected to it) and one phone connected to planet VIP-156. When I try to make call between the phones connected to Linksys I am getting 488 Not Acceptable Here and when I try to reach the phone connected to planet I am getting silence after answer, but the phone can ring so I think that it is a RTP issue. I know that it is caused by the NAT, does anyone know how can I configure this to work appropriately. 488 Not Acceptable Here is almost always a codec issue. ___ --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] Sending Email From A Dial Plan
Untested:
exten = _,1,System(/bin/mail -s \Happy Message: ${EXTEN}\
[EMAIL PROTECTED])
This assumes you can send mail outside of Asterisk from that host.
Damien Gabrielson wrote:
I'm looking for a simple way to send email from a dial plan. I have
searched around quite a bit looking for a solution for this and I'm
surprised that I haven't found anything useful yet other than using
the System() application. I would like to be able to change the
subject dynamically based on ${EXTEN} and the body is not important. I
was hoping to have a one line command from the System() application
without having to write a script or any other dependency. Has anyone
implemented anything like this?
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Re: [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)
The call is not being picked up. Manrique Feoli wrote: 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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Comfort noise support incomplete in Asterisk (RFC 3389).
Luciano Moreira wrote: I trying to setup a outbound trunk with IPSmarx. It's working, but when I make a call, the ring dialtone stills ringing on my side, even after the other side picksup the phone. I got a NOTICE message from Asterisk that I hope you can help me: -- Called [EMAIL PROTECTED] -- SIP/ipsmarx-out-0995f270 is making progress passing it to IAX2/1010-14 -- SIP/ipsmarx-out-0995f270 is ringing -- SIP/ipsmarx-out-0995f270 is making progress passing it to IAX2/1010-14 Aug 16 15:39:21 NOTICE[16215]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230 ipsmarx-out is my outbound route. I got two SIP passing process. So I listen 2 ringtone and when the second ringtone start with a delay I got this NOTICE from asterisk:Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230. I googled this error but could find a fix to this bug. This is not a bug. Contact your provider and tell them to turn off CNG/VAD/Silence Supression as Asterisk does not support this feature. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 'Hosting'
Why make things so much more complicated than they need to be. Asterisk has had support for doing this for ages. The term you are looking for is contexts. Brandon Galbraith wrote: You could use Xen on Fedora Core 6 and virtualize each instance if you feel the need is there. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Matt Riddell (NZ) [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. Why not just use different contexts for each company? Because Asterisk wasn't designed with carrier class features in mind. It was designed for a single enterprise. The dialplan, and config files, start to get very very complicated after you add more than a few companies. Combine that with having to have multiple extensions for a single function (our Queues are accessed by a regular extension but then have to dial another 'virtual' extension so that DUNDi can work out the 'primary' server for a queue) and so on. Anyway, it's becoming unmanagable. Doug. ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] No zap command?
chan_zap won't build if Zaptel isn't installed when you build Asterisk. Rebuild Asterisk after installing Zaptel. Ken D'Ambrosio wrote: Hi, all. I've just set up an Asterisk box -- to the best of my knowledge, no differently than any of the others that I've set up. Only one minor caveat: there's no zap command. Huh? Glancing at the startup, there's no mention of chan_zap, which I assume is partially the reason. However, I'm using -the exact same- zapata.conf, extensions.conf, and zaptel.conf from a different install, so I would imagine it would have been invoked if it were a config issue. Is there a compile-time option that we missed? [And, for the record, no zap errors whatsoever in the log. So it's not like it's trying to load chan_zap.o and failing or anything.] Any ideas would be greatly appreciated... -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Ringing after answered on zaptel
That's kind of useless since progressinband only applies to digital interfaces. Try callprogress=no Brodie Macleod wrote: Try setting: progressinband=no in your sip.conf -Brodie On Monday 14 August 2006 10:20 pm, Don Fanning wrote: Greetings List, I'm having a strange problem with my X100p card still ringing after the call is connected. Any idea on how to solve this? Using latest asterisk (not svn) along with latest zaptel driver. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Macro inside macro
Rushowr wrote: Hey Attilla, thanks for the update. I'm also working on a solution, but unfortunately the system I'm working with needs the separate macros. I'll update the list if anything gets worked out. pbx-1*CLI show application gosub pbx-1*CLI -= Info about application 'Gosub' =- [Synopsis] Jump to label, saving return address [Description] Gosub([[context|]exten|]priority) Jumps to the label specified, saving the return address. pbx-1*CLI -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Macro inside macro
Any reason that you can't set variables before you use Gosub, then access them in the subroutine? Attilla De Groot wrote: On Aug 14, 2006, at 6:39 PM, Eric ManxPower Wieling wrote: Rushowr wrote: Hey Attilla, thanks for the update. I'm also working on a solution, but unfortunately the system I'm working with needs the separate macros. I'll update the list if anything gets worked out. pbx-1*CLI show application gosub pbx-1*CLI -= Info about application 'Gosub' =- [Synopsis] Jump to label, saving return address [Description] Gosub([[context|]exten|]priority) Jumps to the label specified, saving the return address. pbx-1*CLI Already considered this option, but I want to give it some arguments. And that isn't possible with gosub. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Qualify
Jason Parker wrote: I think you misunderstand what qualify is/does. It appears that you believe that qualify=1000 means that it'll send out a qualify packet every 1000ms. This isn't an unreasonable assumption, but it is wrong. The qualify=1000 means that Asterisk will wait 1000ms for the device to respond to the qualify packet. If after 1000ms there is no yes, I'm here packet, then it will be considered UNREACHABLE. Qualify packets are sent out at a set interval, which, as you can see, is 60 seconds. If the device was previously determined to be UNREACHABLE, the qualify packets will then be sent out every 10 seconds instead. One thing to remember, a qualify packet is a SIP OPTIONS packet, not a ping packet. Many phones are very slow in responding to an OPTIONS packet. If the phone got busy doing something like downloading a rinetone or saving a directory entry, the phone may take a while to respond to the options packet. As you can see, you cannot use the qualify option to measure network latency between the server and the SIP device. My main issue with the qualify option is that if even one OPTIONS packet is lost or if the phone is busy doing something and so takes longer then the qualify= is set to, the phone will become UNREACHABLE. In 1.2, chan_IAX2 has a smoother to do some sort of averaging on qualify response times. chan_sip does not have this. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Qualify
Douglas Garstang wrote: Yes, it might be a problem in our situation. We have three Asterisk boxes in a 'cluster'. The sip.conf is identical on all three. In that case, all three of the Asterisk boxes in our cluster are going to send sip options messages to the phones, which is silly. Only the Asterisk box that a phone is registered on needs to send the sip notify messages. The rest are a waste. I'm not sure how we'd work around this. If the phone is not registered to the other two Asterisk servers, how will Asterisk know what IP to send the OPTIONS (qualify) packet to? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Zap difficulties
Curt Shaffer wrote: I am having a weird issue with my zap channel (Digium TDM01B). Randomly it appears that the POTS line is not seeing all of the digits passed. We have to dial a 1 and the area code to call most numbers here, and we get the error that we need to dial a 1 and the area code when dialing this number even though we are dialing it. Also when I dial 8xx numbers it never works (same error). I do have all of those set up as allowed and routing properly from the dial plan and I can test that by switching to a VoIP termination and the calls go through without a hitch. I can also dial these numbers fine if I hook a POTS phone directly to the cable that connects to the Digium card. Asterisk looks as if it is passing the digits, (ZAP/g0/18003569377|120|r) for example. Dial(ZAP/g0/w18003569377|120) This will put a .5 second wait before dialing to allow the telco equipment to get ready to receive DTMF. Have you noticed other issues like, even when calling busy numbers, you hear a ringing tone for about 5.5 seconds before you hear a busy tone? That's because you are using the r option to Dial. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto retry on Busy
RetryDial (and DIALSTATUS) won't work on analog lines.
John Novack wrote:
Also many so-called legacy hybrid PBX switches have had this for many
a year
Hard to compete when well used features that have been around for 20
years are lacking
John Novack
Rushowr wrote:
The reason he might want it is because it's a feature offered by many
POTS and Mobile Telcos. I know that's why I've played with it, the
ITSPs and SIP Termination providers I consult for want to have as many
if not more features to offer than the POTS and Mobile guys.
Cheers,
Rushowr - Sherwood McGowan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent: Friday, August 11, 2006 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto retry on Busy
Why don't you just test for the dial status after the dial command
completes? I don't really see why you want something to keep dialing
until
it gets through, but this would work.
[something]
1,1,Dial(zap/,sip/, etc/whatever, 10)
1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
1,n(LINEBUSY), Wait(30)
1,n,goto(something,1,1)
1,n(OTHER), do something else
Sure it is pretty rough, but the basics are there. Also you might want to
read this: http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Kevin
Noah Silverman wrote:
Hi,
Does anybody have an easy solution for this.
I want something that will keep trying a busy number every 30 seconds
until it gets through.
I've tried retrydial, but can't get it to work.
Any suggestions?
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Re: [asterisk-users] Auto retry on Busy
Also BUSY != BUSY Remember, pretty much any place in Asterisk quotes
are literal.
If you want to test if DIALSTATUS is equal to BUSY you want either:
GotoIf(${DIALSTATUS} = BUSY
or
GotoIf(${DIALSTATUS} = BUSY
Ira wrote:
At 11:54 AM 8/11/2006, you wrote:
Thanks for the suggestion. I can't seem to get it to work.
This is what I put in my extensions.conf
We only have one number that we want to keep trying right now, so I
tried to set it so by calling extension 777, it would start the
system retrying. (The actual number isn't 999 :)
[trunkretry]
exten = 777,1,Dial(${TRUNK}/www1323999},10,)
exten = 777,2,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
exten = 777,3,(LINEBUSY), Wait(15)
exten = 777,4,goto(trunkretry,1,1)
Go read about DIALSTATUS, it's not BUSY you're looking for, it's the
dialstatus BUSY value.
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Re: [asterisk-users] Fast busy signals... Satisfying my curiousity
J. Oquendo wrote: Asterisk Admin calls T1 Provider Asterisk Admin -- T1 customer service -- Do you see 2125551212 dialing in? T1 Cust Svce -- Asterisk Admin -- Nope Asterisk Admin -- T1 Cust Svce -- OK, YOU try calling 2125551212 from both on net and from off net. Where off net is from a line NOT ON THE PROVIDER'S NETWORK. The T1 Customer service won't see the call coming in and so cannot say it's your issue, not our issue -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] MailboxExists not branching to n+101
pbx-1*CLI show application MailboxExists
pbx-1*CLI
-= Info about application 'MailboxExists' =-
[Synopsis]
Check to see if Voicemail mailbox exists
[Description]
MailboxExists([EMAIL PROTECTED]|options]): Check to see if the specified
mailbox exists. If no voicemail context is specified, the 'default' context
will be used.
This application will set the following channel variable upon completion:
VMBOXEXISTSSTATUS - This will contain the status of the execution
of the
MailboxExists application. Possible values include:
SUCCESS | FAILED
Options:
j - Jump to priority n+101 if the mailbox is found.
Ryan Hayward wrote:
Here's the relevent section of my extensions.conf:
### Handle voicemail
exten = _1XX,1,SayDigits(${EXTEN})
exten = _1XX,2,MailboxExists(${EXTEN})
exten = _1XX,3,Playback(vm-nobox)
exten = _1XX,4,Goto(teliax,5013584196,3)
exten = _1XX,103,VoiceMail(b${EXTEN})
exten = _1XX,104,Goto(teliax,5013584196,3)
From what I understands 2,MailboxExits() should branch to 103 if the
box exists, and 3 if the box doesn't. However, no matter if the box
exists or not, it always goes to 3.
If instead of MailboxExists(), I just do VoiceMail(), I get similar
results: The Voicemail() call works if there's a box present, but
whether the mailbox exists or not, it branches to the n+1, instead of
n+101.
Is there something I'm not understanding about priorites, or
extensions that's keeping this from working? I understand what
everything else is doing, and am developing a fairly complicated
extensions.conf, but the branching doesn't seem to work right for me.
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Re: [asterisk-users] can i detect a voice with asterisk ?
Ira wrote: At 07:59 AM 8/10/2006, you wrote: is there a way that asterisk can detect when someone speaks ? Like answering a phone? i dont need speech recognition or anything like that, just something that lets me know that any sound is originating from the other end. Play a recording that says Press 1 to continue over and over. Or stop using analog ports. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Set DID?
Dean Collins wrote:
Is there a command for setting of a DID number?
Eg below I can set callerid
[custom-fromiaxfwd]
exten = s,1,Set(CALLERID(number)=2125316214)
Butw what I would prefer to do is set DID -like this (it doesn't work
[custom-fromiaxfwd]
exten = s,1,Set(CALLERDID(number)=2125316214)
I couldn't find anything in the voip-info commands section so was hoping
for a clue from the list.
You are trying to set the CallerID, not setting the DID. The DID is in
${EXTEN}.
If you want to set the CallerID for calls to the PSTN you must be using
ISDN. If you are using VoIP, then the VoIP server must be using ISDN
(pretty much all of them are).
Your carrier must permit you to set that info. Not all of them do.
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Re: [asterisk-users] Set DID?
Then you would use a Goto(custom-fromiaxfwd,2125316214,1) instead of the
Set(CALLERID
In Asterisk there is no difference between a DID and an extension.
Dean Collins wrote:
Hi Eric,
No I know what I want. I want to set the DID to be 212-531-6214 as my
current provider doesn't send a DID number.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, 10 August 2006 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set DID?
Dean Collins wrote:
Is there a command for setting of a DID number?
Eg below I can set callerid
[custom-fromiaxfwd]
exten = s,1,Set(CALLERID(number)=2125316214)
Butw what I would prefer to do is set DID -like this (it doesn't
work
[custom-fromiaxfwd]
exten = s,1,Set(CALLERDID(number)=2125316214)
I couldn't find anything in the voip-info commands section so was
hoping
for a clue from the list.
You are trying to set the CallerID, not setting the DID. The DID is
in
${EXTEN}.
If you want to set the CallerID for calls to the PSTN you must be
using
ISDN. If you are using VoIP, then the VoIP server must be using ISDN
(pretty much all of them are).
Your carrier must permit you to set that info. Not all of them do.
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Re: [asterisk-users] Ever donate Software to Digium? If you did your a fool.
George Gardiner wrote: Digium is not being given a whole load of money - the investors will want a slice of the company and the future profits. That's how VC funding works. More like selling your soul to the Devil, actually. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Polycom 1.6.7 firmware?
Louis-David Mitterrand wrote: Hello, I am looking for the latest 1.6.7 Polycom firmware? Is it available somewhere? What issues are you experiencing that 1.6.7 fixes? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] PRI Connection in Lima, Peru
On Digital interfaces (PRI, SIP, etc) you are expected to check the value of HANGUPCAUSE and play the correct message to the caller. The telco does not do this for you on these types of interfaces. Carlos Prieto wrote: OK, sorry for not being so explicit. Here is the console output when i try to call no a non-existant number. I don't get the message from the provider telling me the number does not exist. But, if i place a call through an analog line, i got the provider message. -- Accepting AUTHENTICATED call from 201.240.77.46: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (g729|gsm|ulaw|alaw), priority = mine -- Executing Macro(IAX2/599-2, dialout-trunk|1|5622716||) in new stack -- Executing GotoIf(IAX2/599-2, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(IAX2/599-2, user-callerid) in new stack -- Executing GotoIf(IAX2/599-2, 0?report) in new stack -- Executing GotoIf(IAX2/599-2, 0?start) in new stack -- Executing Set(IAX2/599-2, REALCALLERIDNUM=599) in new stack -- Executing NoOp(IAX2/599-2, REALCALLERIDNUM is 599) in new stack -- Executing Set(IAX2/599-2, AMPUSER=599) in new stack -- Executing Set(IAX2/599-2, AMPUSERCIDNAME=Carlos Prieto) in new stack -- Executing GotoIf(IAX2/599-2, 0?report) in new stack -- Executing Set(IAX2/599-2, CALLERID(all)=Carlos Prieto 599) in new stack -- Executing NoOp(IAX2/599-2, Using CallerID Carlos Prieto 599) in new stack -- Executing Macro(IAX2/599-2, record-enable|599|OUT) in new stack -- Executing GotoIf(IAX2/599-2, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(IAX2/599-2, recordingcheck|20060808-125641|1155059801.39) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060808-125641|1155059801.39: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(IAX2/599-2, No recording needed) in new stack -- Executing Macro(IAX2/599-2, outbound-callerid|1) in new stack -- Executing GotoIf(IAX2/599-2, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(IAX2/599-2, REALCALLERIDNUM is 599) in new stack -- Executing Set(IAX2/599-2, USEROUTCID=) in new stack -- Executing Set(IAX2/599-2, EMERGENCYCID=) in new stack -- Executing Set(IAX2/599-2, TRUNKOUTCID=) in new stack -- Executing GotoIf(IAX2/599-2, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf(IAX2/599-2, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf(IAX2/599-2, 1?report) in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp(IAX2/599-2, CallerID set to Carlos Prieto 599) in new stack -- Executing Set(IAX2/599-2, GROUP()=OUT_1) in new stack -- Executing GotoIf(IAX2/599-2, 0?108) in new stack -- Executing Set(IAX2/599-2, DIAL_NUMBER=5622716) in new stack -- Executing Set(IAX2/599-2, DIAL_TRUNK=1) in new stack -- Executing AGI(IAX2/599-2, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(IAX2/599-2, OUTNUM=5622716) in new stack -- Executing Set(IAX2/599-2, custom=ZAP/g1) in new stack -- Executing GotoIf(IAX2/599-2, 0?16) in new stack -- Executing Dial(IAX2/599-2, ZAP/g1/5622716|120|tTrwW) in new stack -- Requested transfer capability: 0x00 - SPEECH * -- Called g1/5622716* -- Zap/1-1 is proceeding passing it to IAX2/599-2 *--* *Channel 0/1, span 1 got hangup request **-- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1)* -- Executing Goto(IAX2/599-2, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(IAX2/599-2, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(IAX2/599-2, outisbusy|) in new stack -- *Executing Playback(IAX2/599-2, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'es') *-- Executing Playback(IAX2/599-2, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'es') -- Hungup 'IAX2/599-2' Here is the console output when calling to an existant number. From time to time, totally random; i got the previous message. -- Accepting AUTHENTICATED call from 201.240.77.46: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (g729|gsm|ulaw|alaw), priority = mine -- Executing Macro(IAX2/599-2, dialout-trunk|1|3623885||) in new stack -- Executing GotoIf(IAX2/599-2, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(IAX2/599-2, user-callerid) in new stack -- Executing GotoIf(IAX2/599-2, 0?report) in new stack
Re: [asterisk-users] Polycom 1.6.7 firmware?
Generally yes, but keep a copy of the old files around just in case. Stephen Murphy wrote: Can you simply replace your current sip.Id and sip.ver files with the latest firware files or is this dangerous? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: August 8, 2006 3:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 1.6.7 firmware? Dean Collins wrote: Yep, but didn't [EMAIL PROTECTED] have a folder to store these files on? Does freepbx? You mean TrixBox? I know they're working on a phone provisioning system, but I thought it was just for Cisco and Grandstreams. Check with the TrixBox guys at http://www.trixbox.org (FreePBX is just a GUI configuration utility. TrixBox is the successor to [EMAIL PROTECTED], i.e. the all-in-one Asterisk-in-a-Can distribution. TrixBox uses FreePBX as part of its management tools). -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 2572 echo can and zaptel yellow alarm
In my experience Yellow Alarm (AIS) on Tellabs indicates that the box does not see a T-1 on one side. marvin horst wrote: Bad card? I wired up another card and got the same result . On the red Rcv In, have you tried swapping out the cable for the opposite cable type. If xfer, change to straight though? Also, have you tested both of the cables to verify they are good? I've confirmed that the cables are good. I tried it on another asterisk system that is a 1.2 version, same result. Both systems have the old T100P digium cards maybe that's the problem. I guess I'll have to purchase the newer cards with the hardware echo can. :( I had hoped to get these tellab cards working with my current hardware. Does anyone have other suggestions? ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] FXS gateway/Channel Bank
Adtran TA750 or TA850 Roger Workman wrote: Can someone recommend a good FXS gateway/Channel bank that will intergrate smoothly with * I need to port over 158 analog lines -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Variables sip redirects and call forward
Check that status of:
${RDNIS} and/or ${CALLERID(rdnis)}) in
/path/to/src/asterisk/docs/README.variables
C F wrote:
First my little Sunday story.
A client of mine with a big factory calls me up that he is trying to
call in to his place because the fire alarm went off. He is dialing
the extension I gave him that will call all the extensions (and worked
before) but after 2 rings he gets a message: The subscriber you are
trying to reach is unavailable (probly from T-Mobile or Cingular). So
I tell him immediately just keep calling back until someone picked up,
I realized that someone must have forwarded their extension (using
local sip forward) to their cell phone, and their cell phone is out of
reach. Sure enough a few hours later when I had a chance to log into
the box, I did my test call and there it was in the CLI calling blah
blah thanks to sip blah at 192.168.24.247 blah blah.
After working with this for around half an hour I just ended up
disabling local forward using the .cfg files from cisco.
So here is my question, is there anyway to determine thru variables
that a phone call is forwarded thru a sip redirect (I think its 302
moved temp)?
Here is my setup:
exten 123 calls a macro that takes a few arguments one of them is the
name of the queue,
the macro in turn calls the Queue, the queue has:
member =sip/1
member = sip/2
member = sip/3
In my case sip/3 was forwarded to an outside number.
so app_queue just followed the sip moved.
Here are my questions:
1. Is there any way to have app_queue ignore redirects?
2. Is there any way to detect in the DP that an extension is called
from a redirect (any varialbes)?
3. While playing around with this problem I tried setting a variable
in the macro that calls the queue using both _VARNAME and __VARNAME
and I was not able to see it anywhere outside the macro that is,
neither was I able to see any of the setvar=VARNAME from sip.conf for
sip/3 in any of the extensions created thru the 302 from sip/3, I also
tried _VARNAME and __VARNAME, is there a workaround to either have the
var from the macro show up everywhere the member it calls end up? or
to have a sip acconts setvar show up even on the channel created from
the 302?
Thank you all.
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Re: [asterisk-users] Setting CALLERID on a residential telco line
Andrea Spadaccini wrote: Ciao Eric, If you had a PRI (not just a T-1) AND your telco permits you to set it. Is there any hope to change the caller-id on a BRI line? Sorry, I was being USA-centric. It's a bad habit to get into. As I understand it, if you have a BRI and your telco allows you to, you can change the outgoing Caller*ID. With PRI, many telcos allow you to set the outgoing Caller*ID. I don't know how often telcos permit this on BRI. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 check the status of a channel
Marcus Carlson wrote: Thomas Artner skrev: Hi! I have two extensions (25 and 26, and so two phones) for one person in an office. I can dial 25 or 26 and always both extensions are ringing. Thats okay! exten = 25,1,Dial(Sip/25Sip/26) exten = 26,1,Dial(Sip/25Sip/26) The problem with this solution is, if the person is talking on one phone and 25 or 26 is called from anywhere, the other phone is ringing. But I would like a busy signal if the person is talking on one of these two phones. How could I do that in the dialplan? I couldn't find something to check whether one of these two channels is busy or not. Any suggestions for me? thx, Thomas ___ --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 This might be what you're seeking; http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail If the phone rings, then the channel IS available. The solution is to disable call waiting on the SIP device. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 2572 echo can and zaptel yellow alarm
The Tellabs cards I used were not configured for ESF/B8ZS when I got them. If you have the Tellabs chassis, try connecting with a serial connection. Here's a copy of the manual: http://www.fnords.org/~eric/tellabs/ It's in PDF format in 2 parts. marvin horst wrote: You can take a regular straight though cable and plug it into the green and the other end into the red. You should get a yellow AIS light on send-in and receive-in. If you don't, then your jacks are probably the issue. I get a yellow AIS light on both send-in and receive-in. Mine is: T1 cross over from Red jack to T110P (Going by the picture on the Wiki). Straight though from Green jack to channel bank. same here I know it's wired properly because you can set mode LPb to 3 (metallic bypass) which bypasses any processing on the echo card. After doing this T100P was communicating with channel bank as before. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Setting CALLERID on a residential telco line
If you had a PRI (not just a T-1) AND your telco permits you to set it. hugolivude wrote: That's what I feared. I could do it if I had a T1 is that right? Thanks, H On 8/4/06, Steven Ringwald [EMAIL PROTECTED] wrote: hugolivude wrote: Redhat 9 Asterisk - 1.2.7 TDM 400 - 1 FXO, 2 FXS I'm using a standard residential PSTN line on my ZAP channel and curious whether I can override the caller ID my telco has for me with one of my choosing. I've tried this: exten = s-ZAP,n,Set(CALLERID(all)=My Name 999-999-999) exten = s-ZAP,n,Dial(Zap/g2/6137451576) but the callee still sees my telco callerid. Have I missed something or does the telco ultimately control CallerID on a residential line? It stands to reason it would, but I'm hopeful I'm wrong!! If it is a POTS line, you cannot change the caller*id. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP_HEADER() read-only
I don't understand what the problem is. If you want to pass a variable
set the variable, but prefix it with __ (2 underscores)
Set(__DNID=${DNID})
Douglas Garstang wrote:
Oh... That's real nice. I was considering using SIP instead of IAX to trunk
calls between Asterisk boxes as IAX has some severe limitations in regards to
passing variables. A few people said 'use SIP!' because you can set the SIP
headers. Looks like that isn't an option!
-Original Message-
From: Vincent Regnard [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 03, 2006 4:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP_HEADER() read-only
There is presently no .write member in the structure declaration for
this function in channels/app_sip.c:
static struct ast_custom_function sip_header_function = {
.name = SIP_HEADER,
.synopsis = Gets or sets the specified SIP header,
.syntax = SIP_HEADER(name),
.read = func_header_read,
};
So I imagine the answer to my question is yes SIP_HEADER is a
read-only
function. There is no implementation equivalent to SipAddHeader() for
SIP_HEADER().
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Re: [asterisk-users] Run a script at certain CLI writes
Bart Fisher wrote: I'm trying to detect when a T1 goes to Yellow or Red alarm. I noticed these events will be displayed on the CLI. What I'd like to do is cause an email to be sent when from a script on these events, but somehow I would need to capture the CLI outputs to detect messages Message are: wct4xxp: Setting yellow alarm on span 1 wct4xxp: Clearing yellow alarm on span 1 Any clues? Wouldn't it be easier to just look at the /proc/zap entries. The ones with the current alarm state should be in cleartext. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232
Pablo Mora wrote: Did you saw my dialplan? I don't think I would have to add r. You never want to add r option to Dial() ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232
Pablo Mora wrote: [outgoing] exten = 0,1,Dial,Zap/g1 exten = 0,2,Hangup exten = 0,102,Congestion You NEVER want Dial,Zap/g1 You If you want to just get an outside dialtone you ALWAYS want a trailing / Dial,Zap/g1/ -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Run a script at certain CLI writes
Eric ManxPower Wieling wrote: Bart Fisher wrote: I'm trying to detect when a T1 goes to Yellow or Red alarm. I noticed these events will be displayed on the CLI. What I'd like to do is cause an email to be sent when from a script on these events, but somehow I would need to capture the CLI outputs to detect messages Message are: wct4xxp: Setting yellow alarm on span 1 wct4xxp: Clearing yellow alarm on span 1 Any clues? Wouldn't it be easier to just look at the /proc/zap entries. The ones with the current alarm state should be in cleartext. Span in ALARM: [EMAIL PROTECTED] zaptel]# cat /proc/zaptel/2 Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 B8ZS/ESF RED 25 TE2/0/2/1 FXSKS 26 TE2/0/2/2 FXSKS 27 TE2/0/2/3 FXSKS 28 TE2/0/2/4 FXSKS 29 TE2/0/2/5 FXSKS 30 TE2/0/2/6 FXSKS 31 TE2/0/2/7 FXSKS 32 TE2/0/2/8 FXSKS 33 TE2/0/2/9 FXSKS 34 TE2/0/2/10 FXSKS 35 TE2/0/2/11 FXSKS 36 TE2/0/2/12 FXSKS 37 TE2/0/2/13 EM 38 TE2/0/2/14 EM 39 TE2/0/2/15 EM 40 TE2/0/2/16 EM 41 TE2/0/2/17 42 TE2/0/2/18 43 TE2/0/2/19 44 TE2/0/2/20 45 TE2/0/2/21 46 TE2/0/2/22 47 TE2/0/2/23 48 TE2/0/2/24 Span NOT in Alarm: [EMAIL PROTECTED] zaptel]# cat /proc/zaptel/1 Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 B8ZS/ESF ClockSource 1 TE2/0/1/1 Clear (In use) 2 TE2/0/1/2 Clear (In use) 3 TE2/0/1/3 Clear (In use) 4 TE2/0/1/4 Clear (In use) 5 TE2/0/1/5 Clear (In use) 6 TE2/0/1/6 Clear (In use) 7 TE2/0/1/7 Clear (In use) 8 TE2/0/1/8 Clear (In use) 9 TE2/0/1/9 Clear (In use) 10 TE2/0/1/10 Clear (In use) 11 TE2/0/1/11 Clear (In use) 12 TE2/0/1/12 Clear (In use) 13 TE2/0/1/13 Clear (In use) 14 TE2/0/1/14 Clear (In use) 15 TE2/0/1/15 Clear (In use) 16 TE2/0/1/16 Clear (In use) 17 TE2/0/1/17 Clear (In use) 18 TE2/0/1/18 Clear (In use) 19 TE2/0/1/19 Clear (In use) 20 TE2/0/1/20 Clear (In use) 21 TE2/0/1/21 Clear (In use) 22 TE2/0/1/22 Clear (In use) 23 TE2/0/1/23 Clear (In use) 24 TE2/0/1/24 HDLCFCS (In use) -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Number of Rings Before Asterisk Takes Over
You do not have Caller*ID service, but Asterisk is configured to wait for Caller*ID information. This information is delivered between the 1st and 2nd ring. Joe Pokupec wrote: Hey All, I'm new to this list. I did some Google searching to find the answer but I couldn't articulate the best keywords. I'm using Asterisk @ Home in a small business environment. When an outside call comes into the system on the PSTN, the caller actually hears 2 rings before Asterisk even kicks in and takes the call. Is there a way to reduce this time? Can I have Asterisk deal with the call immediately? I'm using 4 Digium TDM04B cards to handle 4 POTS lines... -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] cmd DIAL - Who picked up the call?
Koopmann, Jan-Peter wrote:
On Friday, July 28, 2006 3:12 PM Kai Ober wrote:
What about DIAL ( |M(macro-name))
and set the userfield in cdr during execution, ...
Set the userfield to what? That is the entire problem. ${CHANNEL} will give me
something like Zap/10-1. ${BRIDGEPEER} is empty. I would love to see the called
MSN in the port-field something like Zap/10-43 if MSN 43 was called... :-) That
would help enourmously.
Zap/10-43 would indicate that this is the 43rd call (call waiting) on
channel 10. Obviously this would have to be removed to do it the way
you want.
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Re: [asterisk-users] Ringing timer
Zenone wrote: But my question was, is it possible to free the channel if it rings too long? Yes. show application dial in the Asterisk CLI will show you where the timeout goes on the Dial line. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] One way screech or tone
Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] One way screech or tone
IP Phone - Asterisk - PSTN. This would be the Echo Canceler on the Asterisk/Zap - PSTN interface. Bill Gibbs wrote: So would this be the remote end echo can freaking out or the Polycom on the caller side? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] One way screech or tone
Then none of this applies. Bill Gibbs wrote: Ok, in my case it would be my Cisco 3660 since that's what talks to the PRI. It talks sip to my Asterisk box. Thanks! Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone IP Phone - Asterisk - PSTN. This would be the Echo Canceler on the Asterisk/Zap - PSTN interface. Bill Gibbs wrote: So would this be the remote end echo can freaking out or the Polycom on the caller side? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, July 26, 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One way screech or tone Bill Gibbs wrote: Randomly, and this is very hard to debug because it happens so quickly on outbound calls I get a one way screech, it's a steady tone that's very loud. The remote end cannot hear it. You can hear the person talking through the tone. I can't describe it but it's bad enough you have to hang up and call back, and everything then is of course fine since it's so random I have not been able to reproduce it on demand. jbot: Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] New message
Someone connected to the Asterisk console using asterisk -r then typed logger reload then exited the session. Ira wrote: This morning I found this message on my Asterisk Console. Does it mean I should be concerned about the security of my system? -- Remote UNIX connection == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Restarted -- Remote UNIX connection disconnected -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] overlapdial and DID not always working
Sebastian Reitenbach wrote: any idea what I can do? especially why it says it ignores the overlapdial parameter, and why it is accepting them nevertheless? are there any timing parameters to tell asterisk to wait a second longer for the last digit? some rx.. tx.. parameters in the zapata.conf? chan_zap cannot change the overlapdial option on a reload. It can only do it on a unload/load. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Regular expression problem
You are using quotes when you should not be. Notice the double quoting
of -- Executing NoOp(SIP/n-5d23, nothing) in new stack
Benjamin Stocker wrote:
Hi!
What's wrong with this?
exten = s,1,Set(myvar=nothing)
exten = s,2,Set(myvar = $[${CALLERID(num)} : ([a-z]+)])
exten = s,3,NoOp(${myvar})
The regular expression in priority 2 matches, but the result is not
assigned to variable myvar, on the console, I see this:
-- Executing Set(SIP/n-5d23, myvar=nothing) in new stack
-- Executing Set(SIP/n-5d23, myvar = abc) in new stack
-- Executing NoOp(SIP/n-5d23, nothing) in new stack
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Re: [asterisk-users] Voicemail not sent via email
Dean @ INKnBITs wrote: I have setup the voicemail.conf as below, but I not receiving any emails. Any thoughts? voicemail.conf [default] 3002 = 1234,Bob Wright,[EMAIL PROTECTED],,|attach=yes I have also uncommitted the mailcmd=usr/sbin/sendmail -t but that does not work. Check the logs on the Asterisk system. On my system at least, the logs for mail are in /var/log/mail/info, /var/log/mail/warning, and /var/log/mail/error -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Just bought a Polycom 501 - I feel like my GXP-2000 was better...
C F wrote: Feelings are for the ignorant. In any case, if you have trouble pinging your phone then you have something wrong on either your network, or you got a damaged phone. Here is my output from pinging a Polycom 501 while in a conversation with app_voicemail: Ping statistics for 192.168.1.246: Packets: Sent = 100, Received = 100, Lost = 0 (0% loss), Approximate round trip times in milli-seconds: Minimum = 1ms, Maximum = 2ms, Average = 1ms If he has something on his LAN that supports CDP, the phone is prolly trying to get it's VLAN info via CDP. Turn that off in the config file or by using the interface on the actual phone. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 Dial Plan to Play Message
[9507] is the incoming User ID. user=8407 is the outgoing User ID. Do you really want them to be different? Dial() will stop processing of the dialplan until the call ends. Do you really want this? r option to Dial will force a ringing sound to the caller, even if the caller should be hearing a all circuits are busy, or your call cannot be completed as dialed or similar message. Do you really want that? [EMAIL PROTECTED] wrote: Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels. exten = 8407,1,Dial(SIP/8407,80,rt) ; permit transfer exten = 8407,n,Authenticate(9461) exten = 8407,n,Playback(pbx-invalid) exten = 8407,n,Hangup() and in sip.conf [9507] type=friend user=8407 secret=xx ;context=from-sip callerid=8407 host=dynamic nat=yes qualify=yes canreinvite=no dtmfmode=rfc2833 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 Dial Plan to Play Message
You can do it one of two ways: 1) make the SIP device dial a predefined number when the user picks up the phone. You do this in the SIP device. Check the manual for that device for detail on how to do this. It's normally called hotline. In extensions.conf have Asterisk run Authenticate before the Dial() line. 2) Let the SIP device dial as normal, but in the dialplan execute Authenticate before the Dial line. Steve Totaro wrote: You could put the phone in a context such as context=restricted in sip.conf In extensions.conf put a context [restricted] exten = _.,1,Answer exten = _.,2,Authenticate(8675301) exten = _.,3,Goto(whateverdialcontext,whateverexten,whateverpriority) replace Allison's recording for authenticate with your own. Unless I am totally missing what you are trying to do. Thanks, Steve Eric ManxPower Wieling wrote: [9507] is the incoming User ID. user=8407 is the outgoing User ID. Do you really want them to be different? Dial() will stop processing of the dialplan until the call ends. Do you really want this? r option to Dial will force a ringing sound to the caller, even if the caller should be hearing a all circuits are busy, or your call cannot be completed as dialed or similar message. Do you really want that? [EMAIL PROTECTED] wrote: Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels. exten = 8407,1,Dial(SIP/8407,80,rt) ; permit transfer exten = 8407,n,Authenticate(9461) exten = 8407,n,Playback(pbx-invalid) exten = 8407,n,Hangup() and in sip.conf [9507] type=friend user=8407 secret=xx ;context=from-sip callerid=8407 host=dynamic nat=yes qualify=yes canreinvite=no dtmfmode=rfc2833 ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 Dial Plan to Play Message
brandon kruz wrote: youll have to decide what context this goes in either [internal] or [incoming] but i hope you can figure this out yourself here is an idea [internal] exten = s,1,Answer() exten = s,n,Playback(pbx-invalid) exten = s,n,Hangup() *sigh* Playback will BY DEFAULT answer the line. The only time you need an Answer() before a Playback() is if you want a Wait() between them. Doesn't anyone read the docs for the applications they use? For a good time type: show application playback -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 dead-air issues with Digium TE110P and IVR/meetme/internal directory-
Maxx Lobo wrote: An update: I've found that I can leave the TE110P card in the server, unload the module and issue an 'amportal restart' - this brings the IVR/meetme/internal directory voice prompts all back again. So it looks like the issue is directly related to the TE110P module (wcte11xp) in kernel 2.6.9.34-0.2 with CentOS 4.3. Anyone else experience this issue or have any suggestions based on this new information? Do you actually have a line plugged into the TE110P. For some reason I seem to remember reports of audio issues if you have a Digium card in the system, but do not have a line connected to it. I can't cite my source, however. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Question setting up a bat phone extension.
Kai Ober wrote: Eric ManxPower Wieling schrieb: Grandstream seems unable to produce stable firmware. They have tried for *YEARS* and still people have to try many different versions of the firmware to find one that actually works in their environment. okay, i see, thx :) i will try to remember, if i'm ever going to buy an VoIP-Phone. any suggestions for this situation? (i.e. which devices do you prefer) Polycom, Cisco, SIPura/Linksys. I don't like Cisco's firmware licensing, but they are still good phones. Polycoms is the brand of phones we use, SIPura is the brand of ATAs we use. Many people like the Linksys/SIPura phones. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Don't Hit # after 9 to get PSTN line
Turn off 3-way calling on your SIP device. Set the dialplan on your SIP device to not wait 15 seconds after pressing 9. Pablo Mora wrote: Hi all, Iv' got a problem taking lines to call from SIP to PSTN. I have to press # after 9 to get ringtone, otherwise I would have to wait above 15 seconds. [out] exten = 9,1,Dial,Zap/g1/9 exten = 9,2,Hangup exten = 9,102,Congestion The problem occurs when the user doesn't complete the call, and hangup after pressing only 9. If these events occur twice consecutively, Asterisk attempts to native bridge between 2 channels. I think the problem is that # is being used like a transfer trigger. But when I deactivate these feature, I have to wait 15 second after press 9 no get line. What can I do?? What should I do to get line without spend this time? Pablo ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Don't Hit # after 9 to get PSTN line
No. In SIP these features are configured on the SIP device. If you cannot disable three-way calling, or modify the dialplan on your SIP device, then there is nothing you can do to fix the problem. Pablo Mora wrote: I really don't understand what you say. I've been searching in my SIP device (Innomedia 3308), and there isn't any option to disable 3-way calling. Do you refer to sip.conf??? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Question setting up a bat phone extension.
Kai Ober wrote: Has somebody done that with a Grandstream GXP-2000 or a BudgetTone 100/101 ? Has somebody even a list which SIP phones have this funtion? SIPura supports it, Cisco ATAs support it. I assume that Cisco phones support it. I don't know about Grandstream devices since they are banned from our network. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] polycom 601 manual config?
Shaun wrote: Is there not a way to manually configure these phones or at least configure them to use a diffrent tftp server rather than it attempting to ask the dhcp/bootp server? For users at home with dinky linksys/dlink modems you cant set a tftp/bootp server. Of course there is. You do it on the phone. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Question setting up a bat phone extension.
Kai Ober wrote: Eric ManxPower Wieling schrieb: I don't know about Grandstream devices since they are banned from our network. Banned? I didn't try any other devices, but whats wrong with the Grandstreams?? Grandstream seems unable to produce stable firmware. They have tried for *YEARS* and still people have to try many different versions of the firmware to find one that actually works in their environment. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] What is ZapRas used for ?
Angel Diaz wrote: Hi list, What is ZapRas used for ? I would like to use asterisk as a RAS server replacing a Cisco RAS server where users calls to a number directed to asterisk, and here, asterisk answer the data calls and assign an IP address via PPP to calling user. ZapRAS allows Asterisk to act as a dialup server for ISDN DATA calls only. It does not support modem calls. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk and VAD
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... does Asterisk 1.2.7.1 supporting VAD? because i am running my asterisk on VPS and i want to save badwidth. If Asterisk supports VAD (or silence suppression) please tell me how to turn it of! I don't care about bandwidth, I care about sound quality. Asterisk does NOT support VAD. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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: R: [asterisk-users] Called number on ISDN
I believe that with immediate=yes Asterisk does not know what number is
dialed and so that information is not available. Stop using immediate=yes.
Giordano Grandis wrote:
I cannot use it, I have the immediate=yes in my zapata, the extension will be always 's'
Thanks again for all
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Marco Mouta
Inviato: venerdì 14 luglio 2006 15.25
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] Called number on ISDN
Check it ${EXTEN}
On 7/14/06, Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all,
I have an ISDN connection in Italy with MSN. On incoming call how can i
check the dialed number ?
DNID varible could works fine ?
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Re: [asterisk-users] Re: Wrong account code from iax_buddies
Sounds to me that the incoming call is providing the wrong userid/password. voiplist wrote: Anyone have any thoughts on this? On 7/13/06, voiplist [EMAIL PROTECTED] wrote: We have a situation where the wrong account code is being passed from Asterisk to our AGI and then on into the accountcode field in the CDR. Here is the situation, best I can explain it.. We have 3 user records in the iax_buddies table which all come from the same IP address and possibly the same Asterisk server (client side). The accountcode field in the iax_buddies records look like this: name accountcode ipaddr user1 155112.223.225.114 user2 156112.223.225.114 user3 157112.223.225.114 When user1, user2 or user3 terminates a call through the * box the account code doesn't match the accountcode assigned to that user in iax_buddies most of the time. As far as we can tell it only gets mixed up with iax users coming from the same IP. We have lots of other records which show the correct account code on every call. We have searched around and tried for hours to understand how this is possible. All we can come up with is that Asterisk is somehow associating the IP with any of the users names sort of willy nilly regardless of the IAX user the call comes in as. Any help would be appreciated. ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 version: 1.2.9.1 or older?
Martin Joseph wrote: I would love to see a simple explanation of how to update to the latest, including patches. Although I am not using queues, I have wondered about this ever since the change over to SVN, and this seems a good place to ask. The latest release is 1.2.9.1 Anything in SVN is development code. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Dialing timeouts
Doug Lytle wrote: Dan Elder wrote: Hey All, probably missing something really obvious here, but when our users are trying to dial the phone, asterisk timesout really quickly if they don't press the digits fast enough. Is there a global timeout value for dialing See: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DigitTimeout And http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ResponseTimeout These two apps are only for IVR stuff. The timeouts for dialing a call are normally handled by the device. i.e. the SIP phone or ATA, or the zaptel code. For Zaptel see this: /path/to/src/asterisk/channels/chan_zap.c: /*! \brief Wait up to 16 seconds for first digit (FXO logic) */ /* static int firstdigittimeout = 16000; */ static int firstdigittimeout = 2; /*! \brief How long to wait for following digits (FXO logic) */ /* static int gendigittimeout = 8000; */ static int gendigittimeout = 2; -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Text priority labels not working for me
Wes Santee wrote:
Greetings all,
I'm on 1.2.9.1, and I'm trying to get a dialplan working that uses text
labels, but it's not working. For instance, take the following macro
snippet:
[macro-dosomething]
exten = s,1,GotoIf($[${MACRO_EXTEN:1:1} != 1] ? scid)
exten = s,n,Set(MACRO_EXTEN=1${MACRO_EXTEN})
exten = s,n(scid),SetCallerId(${MY_CID})
exten = s,n,Dial(...)
When I call this macro, I get the following:
-- Executing Macro(SIP/1000-66b0, dosomething) in new stack
-- Executing GotoIf(SIP/1000-66b0, 1 ? scid) in new stack
Jul 10 20:05:52 NOTICE[99803]: pbx.c:1753 pbx_extension_helper: No such
label ' scid' in extension 's' in context 'macro-dosomething'
Jul 10 20:05:52 WARNING[99803]: pbx.c:6514 ast_parseable_goto: Priority
' scid' must be a number 0, or valid label
The last log line suggests I can't use labels, but according to
http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities it
shouldn't be a problem.
Am I doing something wrong?
Don't put spaces around the ?
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Re: [asterisk-users] Yet another problem with incoming SIP calls and 407
Wolfgang Zweimueller wrote: Hi all, when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the caller has a username in it's From-Address which also exists in my sip.conf then my system answers with 407 Proxy Authentication Required. If it's nonexistent username then callin works fine! It seems that this is a problem in the SIP implementation of Asterisk and found a few hints on how to resolve this (allowguest=yes, insecure=invite,port etc.). But none of them does help! Can anyone suggest what I else could try? in sip.conf [general] context=INVALID Then put the correct context= line for each sip user/friend/peer. Unauthenticated calls use the options in [general] -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Polycom ACD, Asterisk, Kernel 2.6
Dean @ INKnBITs wrote: I'm trying to build another asterisk server as I'm having a problem with the current one. Unless anybody can tell me how to compile the meetme app? Everything else works fine, asterisk just will not compile meetme?!? (Under kernel 2.4) Meetme will not compile if zaptel is not installed. ___ --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] FXS: No ringtone
Martin Joseph wrote: On Jul 10, 2006, at 1:23 AM, yusuf wrote: Hi all, I am running Asterisk 1.2.7.1, with a Sangoma A101 card in it. I also have 2 Digium FXO cards, and I have premicells connected to the FXO's . Calls come in off the Sangoma E1 cards, from a Philips PABX. The problem I have is that the user, when he dials from his desk phone, does not get any ringtone when he dials a cell phone, which goes over the premicells. So the cell phone will ring, but the user wont hear anything until the cell perosn answers, then everything's fine. But when I try to debug it, I used a sip phone to dial a cell number, that you get ringtone. Yet other calls from the PBX, non cell calls, have ringtone. So when a call uses the E1 anf FXO, I get no ringtone. Has anyone seen this before Oh yeah, what you are talking about is ring back, not ringtone. I think the r option in the asterisk dial command might help you as that forces ringback. The r option seldom fixes ringback issues. Make sure you have /etc/asterisk/indications.conf setup. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Do you need a licence to connect a Ciscohardphone to Asterisk ?
Michiel van Baak wrote: On 16:44, Sat 08 Jul 06, Florian Overkamp wrote: Point is, you do not really need a CH1 or CCME license, you are free to combine the Spare phone with a separate SIP license - the price is identical. It is NOT OK however to use a Spare phone without any license, as far as I am aware. Thanks for the clarification. freakinng licenses they have there :) If you buy a model without the spare in it's name, you have the license to use them right ? How about secondhand phones you get from ebay ? Is my cisco smartnet account enough to run the phone legally ? It's not a spare model (at least that was not in the deal description) If you read Cisco's firmware license it specifically prohibits transfer of the license. So if you buy used phones you still have to buy a new SIP license. This is one of the reasons we went with Polycom instead of Cisco. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] test tone
Outdated, but some of the info may still be current: http://www.tek-tips.com/viewthread.cfm?qid=583069 Edwin Lam wrote: hi folks. does anybody know what's the phone number for SBC Nothern California's 102-type milliwatt test line? (specifically in 415 area code) -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Polycom Soundpoint IP 301 w/ MGCP
There are different versionsof the polycom phones. Depending on the actual part number it can come with MCGP, SIP, or H323. Polycom does not support customer migration from one protocol to another. Get the version with the SIP firmware. Jim Freeze wrote: Hi I was about to order a polycom 301 when I noticed that the VoIP protocol is listed as MGCP and not SIP, as with the 501. First, what is MGCP? And, will the 301's work seamlessly with Asterisk and the other 501 phones that I have? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Dial Macro timeout fails
This has been my experience as well. I also posted the issue to this mailing list, but has not responses. I have not come up with a workaround. If I have time I'll try to write up a bug report, but it will be a while. You are welcome to document the issue with as much detail as you can and post it to bugs.digium.com. David wrote: To add to the mystery, if the cell phone answers and presses 1 as requested, the logs don't register priority 1,1 being executed. It is as if the macro has prematurely aborted. David David said: I just downloaded, compiled and installed Asterisk 1.2.9.1. I did this specifically to get the Dial M(x^y) feature so that I could implement call completion confirmation over IAX2 channels (not available in 1.0.7). The problem is that the call is always completed--even without the required user input. The problem seems to be related to the response timeout. Macro priorities i,1 or t,1 are never executed. Here is what I have in extensions.conf: -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] '500 Internal Server' Error on SIP NOTIFY
Yes. It does not seem to cause any problems. Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address DEFANGED_uri=sip:[EMAIL PROTECTED];user=ip DEFANGED_priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] M() option to Dial
I'm using the M() option to Dial() and having problems. In the
following dialplan example ANY digit exits the macro. When the callee
presses 1 the Noop(Reset AbsoluteTimeout(0)) does not get run. Does
anyone have any ideas as to what I'm doing wrong? Asterisk 1.2.x
[extensions]
exten = 2998,1,Dial(Zap/1/5551212,,wM(answer-confirmation^20))
[macro-answer-confirmation]
exten = s,1,Noop(Set AbsoluteTimeout(${ARG1})
exten = s,n,Background(/etc/asterisk/call-from-campground)
exten = s,n,Goto(2)
exten = 1,1,Noop(Reset AbsoluteTimeout(0))
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Re: [Asterisk-Users] when I press transfer - blind - 700 . The user is not able to hear what extension the call was parked on
sdgesa gaeharth wrote: I have blindxfer = #1 set in features.Doesn't this means #1 is the same as transfer - blind, correct? Both are blind transfers.. Is so, why when I transfer using #1 do I hear what extension the call was parked at but not transfer - blind? #1 is, for whatever reason, doing a supervised transfer. You do NOT get to hear the called party in a blind transfer. If you hear the called party when you do a transfer then it is a supervised transfer, not a blind transfer. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] when I press transfer - blind - 700 . The user is not able to hear what extension the call was parked on
Matt wrote: Interesting, I have #2 setup to do blind transgfers, and if I do a #270 it tells me the number seven one and then hangs up on me and the user is left on park 71. Maybe Asterisk knows that doing a blind transfer to park a call is a silly and pointless thing to do and does a supervised transfer instead? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Soekris net4801 and IAXy dhcp issue
Juan Luis Moyano wrote: Hi all, I have a Soekris net4801-50 board with OpenBSD 3.9 where I've configured a dhcp server and tested it with a regular PC connected directly via a crossover cable with success. The problem comes when I try to connect my IAXy device instead of the PC. I can see with 'tcpdump -nettti sis1' that the IAXy isn't sending any packets to the dhcp server. I thought my IAXy was bad but then I configured a second dhcp server with the exact same config file and the IAXy worked right out. So I don't have a clue of what could be happening. Please shed me some light on this issue. Thanks in advance. I've suspected for a while that the IAXy does not use DHCP, but uses a similar, older protocol called BOOTP. It could not hurt to try and enable BOOTP on your DHCP server (ISC DHCP server supports 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] Echo and crackle
Mojo with Horan Company, LLC wrote: I will agree that switching to the TDM card significantly helped my echo and sound quality, I would take a second to point out that interrupt sharing on your * server might cause crackling-like noises. Try lspci -vb and cat /proc/interrupts to see if you discern any hardware using the same irq the x101p is. lspci does not show the IRQs *after* ACPI is enabled. /proc/interrupts does. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Vincent Delporte wrote: Thanks Noah for the help, but... no go :-/ From: Noah Miller ONE: You should answer an incoming zap line before doing anything with it, so do this: exten = s,1,Answer exten = s,2,Dial(Zap/2/014XX) When I try this, instead of using the Zap/2 interface to ring the other number, Asterisk goes off hook and I hear some kind of static: You have a problem unrelated to what you are trying to do. Fix the problem with dialing out of Zap/2 first. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
