[Asterisk-Users] DTMF Talk off
Hello all,
I have seen some chatter again about DTMF. I see most of the talk about DTMF
around not being able to get an external IVR to recognize digits, not a big
issue for me at this time but sill interesting. My issue though, is with
talk off on a zap channel. It seems to be getting worse or maybe my patience
is thinning. All my calls go out and come in pstn through an FXO as I do not
have high speed available here at home. My Current setup is:
Phone--PAP2-- * ---PSTN---Voip number to * at another location(that has
high speed)---send to VoIP provider
I read a post about talked about the length of the DTMFish sound. I also
remeber seing something about relaxdtmf being set to something other than yes
or no, so I looked in chan_zap.c and found relaxdtmf in many places but it
looked to my inexperienced eye that it could only be set to 'yes' or 'no',
but i did find some mention of tonlength (at line 10858)
followed that to zaptel.c (line 3357) where it said :
if ((tdp.dtmf_tonelen 4000 ) || (tdp.dtmf_tonelen 10 ))
return -EINVAL
Which I am guessing means unless the dtmf is between these 2 values ignore it.
Any ideas what might happen if i increased the minimum time value? if this is
indeed what this is referring to?
Zapata.conf:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
busydetect=yes
busycount=6
echocancel=128
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=0
immediate=no
context=default
signalling=fxs_ks
channel = 1
same for channel 2
zaptel.conf:
loadzone = us
fxsks=1
fxsks=2
extensions.conf:
exten = s,1, NoOp(${CALLERID} time ${DATETIME});
exten = s,2, Dial(sip/677sip/666,30,tT);
exten = a bunch of stuff to do with agi look ups and voicemail
leave/retrieve
All very basic and works like a charm except for the talk off.
Thanks in advance to all,
John M
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Re: [Asterisk-Users] asterisk on AMD 64 BIT
I have 2 servers currently running 64 bit SuSE 10.x on AMD Opteron processors both of which are working very well. John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 On Monday June 12 2006 1:39 pm, Kanishka Somaratne wrote: Hey Does asterisk works well on an AMD 64 bit processor server. are there any issues with this ? Regards Kani -- ___ --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] no audio between sip channels * 1.2.6
Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the other and no sound. Okay must have screwed something up. checked sip.conf all looked good. Okay good time to go to 1.2.6, still no audio. All phones ring and answer but no audio. the last thing that apears on the console is attempting native bridge of sip/677- and sip/699- below is a debug of a call and sip.conf. Each channel on the PAP2's is set to a different port 5060 through 5063. I can call in to any phone and all is good, use any phone to call to POTS line and back in on second POTS line and all is good. I have been looking through the archive of the mail list that I keep and have not found anything to fix my problems yet. i have transfered the registration of both PAP2's to a 1.2.0 system that I have and everything works as it should. moved 1.2.0 configs to 1.2.6 box and again no audio between sip channels. *CLI sip debug SIP Debugging enabled *CLI -- SIP read from 192.168.1.200:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: John Millican sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 235 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 143361 143361 IN IP4 192.168.1.200 s=- c=IN IP4 192.168.1.200 t=0 0 m=audio 16410 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 12 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.1.200 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.1.200:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941;received=192.168.1.200 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0 To: sip:[EMAIL PROTECTED];tag=as0767a869 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=6e91851e Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '677' -- SIP read from 192.168.1.200:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0 To: sip:[EMAIL PROTECTED];tag=as0767a869 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: John Millican sip:[EMAIL PROTECTED]:5060 User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 0 --- (10 headers 0 lines)--- -- SIP read from 192.168.1.200:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a3883dd1 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=677,realm=asterisk,nonce=6e91851e,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=121d27cf19808e8a097930f0f969d3d7 Contact: John Millican sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 235 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 143361 143361 IN IP4 192.168.1.200 s=- c=IN IP4 192.168.1.200 t=0 0 m=audio 16410 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 12 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.1.200 : 5060 (non-NAT) Found user '677' Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.200:16410 Found description format PCMU Found description format NSE Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 699 in pap2 (domain 192.168.1.10) list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT) to 192.168.1.200:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a3883dd1;received=192.168.1.200 From: John Millican sip:[EMAIL PROTECTED];tag=f250a44bc61492b1o0
Re: [Asterisk-Users] Asterisk on hosted server
On Friday March 17 2006 8:07 am, Can2002 wrote: I'd planning on install Asterisk on a hosted Linux box we're setting up. The hosting provider that seems to offer the best deal can install either Debian 3.1 or SUSE 9.x or 10.0 in either 32 or 64 bit editions (running on AMD 64 bit). My experience has been gained on RedHat to date, but I do have some SUSE experience. I assume I'll have no problems running Asterisk on SUSE, but I'd appreciate any recommendations. Also, should I be safe on a 64 bit distribution? Cheers, Chris Chris, I have been away for a while so not sure if you have gotten many answers on this yet but... I am running * 1.2.4 on an AMD Opteron 165 dual core with SUSE 10.x 64 bit and all is working very well. John M ___ --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] linksys pap2 automatically connect on liftinghandset
On Friday January 13 2006 10:14 pm, James Harper wrote: The best I can do so far (which appears to be a bit of a hack) is (:0S0), which says to add a '0' to the start of the string and dial immediately. This gives asterisk an extension dialled of '0', which isn't the 's' that i'd hoped for, but is a good start! (S0) by itself doesn't work, nor does (:S0). Any other suggestions? Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Saturday, 14 January 2006 13:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] linksys pap2 automatically connect on liftinghandset Is there a way to configure the linksys pap2 to automatically connect to asterisk on lifting the handset (presumably into the 's' state)? Asterisk would then be listening for DTMF tones to figure out what to do rather than having to put a dial plan into each pap2. I think the pap2 is pretty much the same inside as a few of the sipura boxes so the same thing might work if anyone knows... Thanks James Create an extension that will catch what ever number you put into the PAP2 as (SO:Extension_in_Asterisk) This provides a hot line setup per the documentation for a SPA-3000 which I believe has the same dial plan execution as the PAP2, at least all the stuff i did on one worked on the other. John M ___ --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 Jobs
On Sunday January 08 2006 12:23 pm, Kerry Garrison wrote: Asterisk drastically lowers barriers of entry in the field of commercial telephony systems. Besides, the wiki, the mailing list and the IRC channels make it relatively easy to get started with the system. This no-pointy-clicky no-brainer interface actually allows you to gain more in-depth knowledge about telephony and VoIP. Have your worked with any other PBX system? Learning Asterisk is extremely product-centric. Knowing how to create a dialplan in Asterisk does not necessarily translate into how you create a dialplan on Call Manager or 3Com NBX, or TalkSwitch, or Panasonic, or Toshiba, or Mitel, or anything else. Adding a phone and extension is different on each system, etc etc etc. The advantage of Asterisk is that you can pull obsolete hardware out of your junkyard and get a system up and running and begin learning general telephony and voip methods and terms. I'm not sure why you feel editing config files manually helps your learn faster than using an interface such as AMP. I totally disagree with that, if you don't have to learn all of the syntax all up front and you have an easy means of doing 100% of your configuration through an interface, you will learn how to setup and manage a system much faster. Kerry Garrison I disagree. In my opinion if one learns how to do everthing only through a GUI they are not learning what is actually happing. If you are unaware of what is going on under the hood you will have far greater difficulty finding the problem when something goes wrong. The GUI says all is great but it doesn't work, Oh my god what do I do now. Having said that i go back to an earlier post about different people learning in different ways. this is very true, but it requires more work to learn the underpinnings after learning the GUI. If you are forced to learn the inner working first you can then move to the GUI and have a better idea of what went wrong when it does. I will consent that it is easier to learn how to set something up via a GUI but I don't feel that is the best way. My 2 cents. John Millican ___ --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 To Asterisk/POTS - Hardware Setup Questi on
see bottom post On Wednesday December 21 2005 1:32 pm, Colin Anderson wrote: Um, not trying to be a smartass, but a simple 2 way splitter like the one you get in the dollar store would do the trick nicely. Then you could just plug in a POTS phone and turn the ringer off. Don't think it would suck too much voltage so your FXO card shouldn't notice. hth -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 21, 2005 11:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] New To Asterisk/POTS - Hardware Setup Question Regards to All, I recently setup an Asterisk system ([EMAIL PROTECTED]) and it works like a charm so far. It is in a SOHO behind another Linux iptable NAT firewall with no problems. Hopefully this isn't too dumb a question, and its the right place to ask it. The situation is that at this time I have only one incoming PSTN line which I have not yet hooked up (I have a single port FXO wildcard arriving soon for test purposes) which I would like to have available whether the server is available or not. I'm thinking that a Sipura or Grandstream analog adapter with PSTN passthrough is the solution, but I'm not sure, as I'm new to the whole PBX/POTS system. Everything I've seen with passthrough is also a router/gateway. Is that necesary and will it work or is there a better solution? For example, we have regular power outages here at my location lasting anywhere from 1 minute to two hours and if the system is down I would like to still have access to local 911 as well as other local numbers. The obvious thing to do is just unplug one of the phones and plug it directly into the POTS line, but I'm hoping there is a product available that will work with both Asterisk and allow passthrough that will not only transparent, but be less expensive than setting up a UPS system that will hold the server up for an hour or so. A UPS to hold up the adapting device and phone for an extended period would be far cheaper, I think. TIA for any replies. Regards, John C. In my opinion the solution of a sipura 3000 would be the better of the 2 options you mention. this will allow you to have a receptionist phone plugged into it that will work at all times so there will not be any confusion about when power is on use this phone, when power is out use that phone. somewhat compensates for panic factor. Now this will depend on number of extensions in the SOHO. If this is the only extension all is good. Remember that without a ups on the * box when the power goes out all calls end rather abruptly and that dirty shut downs can be a very bad thing. John M ___ --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] WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data.
On Saturday November 26 2005 1:41 pm, John Millican wrote: On Saturday November 26 2005 1:26 pm, John Millican wrote: Hello all, I have just upgraded to 1.2 from 1.0.9 and am receiving and placing calls as expected. I have been trying to get atxfer working and am getting the error message: WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data. whenever I try a transfer. In features.conf: [general] parkext = 700; parkpos = 701-720; context = parkedcalls; ;parkingtime = 45; transferdigittimeout = 3; courtesytone = beep; xfersound = beep; [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0; Disconnect automon = *1 ; One Touch Record atxfer = *2; Attended transfer [applicationmap] testfeature = #9,callee,Playback,tt-monkeys; in extensions.conf [globals] DYNAMIC_FEATURES=automon#blindxfer#disconect#atxfer#testfeature in CLI when attempting a transfer: SIP/677-8544 answered Zap/1-1 -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Playing 'pbx-transfer' (language 'en') Nov 26 13:13:33 WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data. -- Playing 'beeperr' (language 'en') -- Stopped music on hold on Zap/1-1 and then the channels are joined again as if nothing had happened. I googled for the error message and searched voip-info.org but no results on either. Sorry for the second post but thought I should add some info. setup is: PSTN --- X100P in asterisk box - Linksys PA2-NA phone1 on port 1 and Phone2 on port 2 the linsys is set to g711 ulaw with inband signaling I am trying to transfer an incoming call from phone1 to phone 2 Okay let me try once again. When I attempt a transfer either blind or attended i get the transfer prompt and then dial tone as I should. Then what happens is when I press a digit the dial tone may or may not go away. If I repeat that first digit I can sometimes get the dial tone to go away and asterisk accepts the remaining digits for the transfer without problem and the transfer happens. I have tried increasing the dtmf playback level in the PAP2 from -16db all the way up to 0db. This has not made any noticeable difference in detection. I have also increased the DTMF playback length from .1 to .3 again no success. Any help would be greatly appreciated. Thank You John Millican ___ --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] WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data.
Hello all, I have just upgraded to 1.2 from 1.0.9 and am receiving and placing calls as expected. I have been trying to get atxfer working and am getting the error message: WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data. whenever I try a transfer. In features.conf: [general] parkext = 700; parkpos = 701-720; context = parkedcalls; ;parkingtime = 45; transferdigittimeout = 3; courtesytone = beep; xfersound = beep; [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0; Disconnect automon = *1 ; One Touch Record atxfer = *2; Attended transfer [applicationmap] testfeature = #9,callee,Playback,tt-monkeys; in extensions.conf [globals] DYNAMIC_FEATURES=automon#blindxfer#disconect#atxfer#testfeature in CLI when attempting a transfer: SIP/677-8544 answered Zap/1-1 -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Playing 'pbx-transfer' (language 'en') Nov 26 13:13:33 WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data. -- Playing 'beeperr' (language 'en') -- Stopped music on hold on Zap/1-1 and then the channels are joined again as if nothing had happened. I googled for the error message and searched voip-info.org but no results on either. Thank you for any help, John Millican ___ --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] WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data.
On Saturday November 26 2005 1:26 pm, John Millican wrote: Hello all, I have just upgraded to 1.2 from 1.0.9 and am receiving and placing calls as expected. I have been trying to get atxfer working and am getting the error message: WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data. whenever I try a transfer. In features.conf: [general] parkext = 700; parkpos = 701-720; context = parkedcalls; ;parkingtime = 45; transferdigittimeout = 3; courtesytone = beep; xfersound = beep; [featuremap] blindxfer = #1 ; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer [applicationmap] testfeature = #9,callee,Playback,tt-monkeys; in extensions.conf [globals] DYNAMIC_FEATURES=automon#blindxfer#disconect#atxfer#testfeature in CLI when attempting a transfer: SIP/677-8544 answered Zap/1-1 -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Playing 'pbx-transfer' (language 'en') Nov 26 13:13:33 WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data. -- Playing 'beeperr' (language 'en') -- Stopped music on hold on Zap/1-1 and then the channels are joined again as if nothing had happened. I googled for the error message and searched voip-info.org but no results on either. Sorry for the second post but thought I should add some info. setup is: PSTN --- X100P in asterisk box - Linksys PA2-NA phone1 on port 1 and Phone2 on port 2 the linsys is set to g711 ulaw with inband signaling I am trying to transfer an incoming call from phone1 to phone 2 thanks again John Millican ___ --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] match a set of numbers in GoToIf against a variable
Hello all,
Okay when you are done laughing at the simplicity of this question could
someone show me please what I have wrong in the following statement?
GoToIf($[${numdial} != [1-9] ]?15:3);
What this is supposed to do is if numdial is not a single digit from 1 to 9
inclusive goto 15, if it is a singledigit from 1 to 9 inclusive goto 3.
Should be pretty simple but not working for me, always goes to 15. from what
I read on the wiki this should work, but obviously I must have read it wrong.
I can get this to work:
exten = 1,2, GoToIf($[${numdial} = 1 ]?15:3);
exten = 1,3, GoToIf($[${numdial} = 2 ]?15:4);
exten = 1,4 ,GoToIf($[${numdial} = 3 ]?15:5);
...
exten = 1,15, do other stuff
but I don't want to put this in 9 times.
Thanks in advance,
John M
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Re: [Asterisk-Users] match a set of numbers in GoToIf against a variable
On Friday October 14 2005 8:26 pm, Samy Antoun wrote:
--- John Millican [EMAIL PROTECTED] wrote:
Hello all,
Okay when you are done laughing at the simplicity of
this question could
someone show me please what I have wrong in the
following statement?
GoToIf($[${numdial} != [1-9] ]?15:3);
What this is supposed to do is if numdial is not a
single digit from 1 to 9
inclusive goto 15, if it is a singledigit from 1 to
9 inclusive goto 3.
Should be pretty simple but not working for me,
always goes to 15. from what
I read on the wiki this should work, but obviously I
must have read it wrong.
I can get this to work:
exten = 1,2, GoToIf($[${numdial} = 1 ]?15:3);
exten = 1,3, GoToIf($[${numdial} = 2 ]?15:4);
exten = 1,4 ,GoToIf($[${numdial} = 3 ]?15:5);
...
exten = 1,15, do other stuff
but I don't want to put this in 9 times.
Thanks in advance,
John M
John,
Will this meet your needs:
exten = _Z,1,Will match any single digit from 1 to 9
exten = i,1,will match enything else (invalid)
thanks for the reply but i am not sure if that will work. Let me try and see.
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Re: [Asterisk-Users] match a set of numbers in GoToIf against a variable
On Friday October 14 2005 8:57 pm, John Millican wrote:
On Friday October 14 2005 8:26 pm, Samy Antoun wrote:
--- John Millican [EMAIL PROTECTED] wrote:
Hello all,
Okay when you are done laughing at the simplicity of
this question could
someone show me please what I have wrong in the
following statement?
GoToIf($[${numdial} != [1-9] ]?15:3);
What this is supposed to do is if numdial is not a
single digit from 1 to 9
inclusive goto 15, if it is a singledigit from 1 to
9 inclusive goto 3.
Should be pretty simple but not working for me,
always goes to 15. from what
I read on the wiki this should work, but obviously I
must have read it wrong.
I can get this to work:
exten = 1,2, GoToIf($[${numdial} = 1 ]?15:3);
exten = 1,3, GoToIf($[${numdial} = 2 ]?15:4);
exten = 1,4 ,GoToIf($[${numdial} = 3 ]?15:5);
...
exten = 1,15, do other stuff
but I don't want to put this in 9 times.
Thanks in advance,
John M
John,
Will this meet your needs:
exten = _Z,1,Will match any single digit from 1 to 9
exten = i,1,will match enything else (invalid)
thanks for the reply but i am not sure if that will work. Let me try and
see. ___
Well just for finality, I ended up using len
GoToIf($[${len(${numdial})} = 1 ]?15:3);
works like a charm.
John M
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[Asterisk-Users] DTMF detection
Hello all, yes there is a lot of information about this on the wiki and in past posts on this list but have not found anything that has solved my problem. setup is: phone--PAP2-na--asterisk v1.0.9(in house on local subnet dtmf is inband)---PSTN---Telisipasterisk box at colo v1.0.9 VoIP only. I have only access to dial up so can not go VoIP out of the house. In extensions.conf on colo * i have some logic that based on callerid lets me hit a single digit to get to DISA, this work every time. the problem is that when i enter a number for DISA to dial i get duplicate digits, example i enter 6037862111 and disa tries to dial 6003778621. I have tried setting relaxdtmf=yes in sip.conf with no luck. I have read on the wiki that RFC2833 should work, but alas its a no go. I am also using ulaw which should not be distorting the dtmf through compresion, correct? Also with RFC2833 it should not matter? Everything works great otherwise. sip.conf for colo * is posted below: [general] context=telasip port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all; First disallow all codecs allow=ulaw register = username:[EMAIL PROTECTED] [telasip] type=peer username=* fromuser=* authname=* secret=* host=gw3.telasip.com context=default dtmfmode=RFC2833 disallow=all allow=ulaw canreinvite=no nat=no Thanks in advance for any help John Millican ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk forwarding confirmation?
Hi; I've been using Asterisk for a few months now, and I have run into
an interesting issue that I thought someone else in the community may
have run into:
I have an Asterisk install set up to receive helpdesk calls, route
them to several IAX extensions and an extension which is simply a
forwarded call over the POTS to a cellphone, so that if no one is
logged into their IAX extensions for whatever reason, the call would
go to a cellphone. Ideally, I'd like it to move onto the next part of
the dialplan if the cellphone isn't answered.
Unfortunately, it turns out that most (if not all cellphones) have
voicemail, which appears to Asterisk as though it had connected with a
person, and it then connects the call. I had wanted to put in a small
AGI application (or something similar) which asked for a single
keypress to confirm that someone had actually picked up the phone
call, but it seems as though using an AGI script would simply prompt
the caller. Has anyone else had this sort of problem, and is there a
way around other than creating call files and attempting to connect
them with the incoming call?
Thanks,
Jeff Buchbinder
Might be a cluedge work around but it seems to me you could ask the user for a
key press and set a var based on receiving the keypress or not. Something to
the effect of:
exten = x,x, Dial(cell_phone,20);
exten = x,x, Read(var|voice_prompt|1);
exten = x,x, GoToIf($[${var} = 9]?extension if key pressed:extension if key
not pressed);
Voice prompt should say blah blah press 9 to accept this call.
Read wiki for the Read and the GoToIf commands.
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Re: [Asterisk-Users] IAX TO IAX call between two registered servers
[snipping] I get the following message on home: Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read: Call rejected by 69.xxx.xxx.xxx: No authority found and get this message on away Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read: Rejected connect attempt from 165.xxx.xxx.xxx home iax.conf [away] type=peer username=away auth=plaintext secret=x host=dynamic context=pap2 dissallow=all allow=ulaw [more snipping] away iax.conf [home] type=peer user=home secret=x host=dynamic context=default [home-in] type=user username=home secret=x context=default [final snipping] any suggestions would be greatly appreciated. Thank you, John M OK, I think your problem is simple, at least I hope so ;-) Specifically, I think you have the wrong syntax in your Dial command. You are dialing: exten = _998, 1, Dial(IAX2/x:[EMAIL PROTECTED]); When i did iax2 debug the above dial string placed the correct username where it should be. the wiki (I thought) said that dial should be Dial(technology/password:[EMAIL PROTECTED]) I will have to check again. The above says (to me) to use channel IAX2, username=x, password=home, in context away (in the local iax.conf). I doubt that's what you want! I think you want to dial extension x, with username home (or remote context home, which isn't place to put it), at local context away. So, change your Dial string to: exten = _998, 1, Dial(IAX2/[EMAIL PROTECTED]/x); That might fail too, for the following reasons: In your Dial command, you are specifying user home, calling via your context away (in home's iax.conf). So far, so good. However, in the away context (on home's machine), you are setting the username to away. There is no context away in the iax.conf on the away machine. Perhaps (I'm not sure), the username=away is the line that will interfere with your authorization (after making the change to the Dial command as noted above). Two suggestions: 1) Delete the line that says username=away in the away context in the iax.conf on home. 2) Change it to be username=home (in which case you can change your Dial command to be just IAX2/away/XXX Thank You 1 and 2 did the trick worked and can now make unauthenticated calls at least. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX TO IAX call between two registered servers
Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees us as 69.xxx.xxx.xxx:4569 on home: Registered IAX2 to '69.xxx.xxx.xxx', who sees us as 165.xxx.xxx.xxx:4569 When i place a call from home to away: exten = _998, 1, Dial(IAX2/x:[EMAIL PROTECTED]); I get the following message on home: Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read: Call rejected by 69.xxx.xxx.xxx: No authority found and get this message on away Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read: Rejected connect attempt from 165.xxx.xxx.xxx home iax.conf [away] type=peer username=away auth=plaintext secret=x host=dynamic context=pap2 dissallow=all allow=ulaw [away-in] type=peer auth=plaintext secret=x host=dynamic context=pap2 dissallow=all allow=ulaw [away-out] type=peer secret=x username=away host=dynamic disallow=all allow=ulaw away iax.conf [home] type=peer user=home secret=x host=dynamic context=default [home-in] type=user username=home secret=x context=default [home-out] type=peer secret=x username=home host=my.domain.com any suggestions would be greatly appreciated. Thank you, John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX TO IAX call between two registered servers
On Monday August 08 2005 7:06 pm, Carlos Chavez wrote:
On Mon, 2005-08-08 at 18:42 -0400, John Millican wrote:
Hello all,
I know this has been covered on list but can not find the answer I need,
lots of references to no authority found, but none with an answer.
I have two * servers, one behind firewall with nat the other on a dmz
with nat. Both servers register with each other successfully.
home is today's CVS-HEAD
away is Asterisk 1.0.7
on away: Registered to '165.xxx.xxx.xxx', who sees us as
69.xxx.xxx.xxx:4569 on home: Registered IAX2 to '69.xxx.xxx.xxx', who
sees us as
165.xxx.xxx.xxx:4569
When i place a call from home to away:
exten = _998, 1, Dial(IAX2/x:[EMAIL PROTECTED]);
I guess what you are trying to do here is dial 998 and then the remote
extension number? If so your extension shoud be something like:
exten = _998.,1,Dial(IAX2/x:[EMAIL PROTECTED]/${EXTEN:3})
I was actually just trying to get it to fall into the default context which is
set up as follows
[default]
exten = s,1, agi,voicemail.cpp|${CALLERIDNUM}; does a db lookup
exten = s,2, GoToIf($[${MAILUSER} = 0]?5:5);
exten = s,3, GoToIf($[${MAILUSER} = 1]?4:5);
exten = s,4, HasNewVoicemail([EMAIL PROTECTED]:INBOX)
exten = s,5, Dial(sip/577,20); Ring the phone on the sipura
exten = s,6, GoTo(myvoicemail,9002,1);
exten = s,105, GoTo(myvoicemail,8002,1);
so that the phone on the sipura should ring since voicemail.cpp will not find
a listed calleridnum. am i totally missing something?
I get the following message on home:
Aug 8 18:31:03 WARNING[20598]: chan_iax2.c:6820 socket_read: Call
rejected by 69.xxx.xxx.xxx: No authority found
and get this message on away
Aug 8 18:32:42 NOTICE[16923]: chan_iax2.c:5448 socket_read: Rejected
connect attempt from 165.xxx.xxx.xxx
home iax.conf
[away]
type=peer
username=away
auth=plaintext
secret=x
host=dynamic
context=pap2
dissallow=all
allow=ulaw
[away-in]
type=peer
auth=plaintext
secret=x
host=dynamic
context=pap2
dissallow=all
allow=ulaw
[away-out]
type=peer
secret=x
username=away
host=dynamic
disallow=all
allow=ulaw
away iax.conf
[home]
type=peer
user=home
secret=x
host=dynamic
context=default
[home-in]
type=user
username=home
secret=x
context=default
[home-out]
type=peer
secret=x
username=home
host=my.domain.com
any suggestions would be greatly appreciated.
Thank you,
John M
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Re: [Asterisk-Users] Cepstral
what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? IIRC, you can download everything you need to make the thing talk, including a voice like David. It works exactly like it will when you buy a license except there is some kind of crippling until you install the license key. I don't remember if this is a statement made by the voice each time or a time out. FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. So much depends on what you are trying to do. I just wanted to have a way to allow asterisk to talk in a demo, to show the concept. Unfortunately, showing a talking server with Cepestral's David is little like showing a prototype website: people don't always have the imagination (like we all do here :) to see what this would be like when actually done (or using a better voice in this case). I have the emily voice and she sounds much like the marine weather station reports. the crippling is just a message that says it is an unregistered version or the like. Yes you can absolutely tell that it is speech synthesis but it is understandable. You can fiddle with the settings, in the readme this is explained, and make it sound a little better. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
I have been reading about Cepstral, their voices and the Digium partner agreement with them. I see where they sell the voices and the licenses for them, but what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? Strangely, the Cepstral web site does not explain this... Can someone shed some light? Thanks... I have been using cepstral for a while now. Swift is the old name(I believe) for cepstral and is placed in the /install_dir/bin directory when you unpack the cepstral download. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? Try this in zapata.conf for fun: busydetect=yes busycount=6 Let us know if it makes a difference. I have added this to zapata.conf. Will let you know what happens. can you tell me why this might help or point me to a wiki/google? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Retrieving dtmf, passing to shell, and getting the result
I have my asterisk server up and running on OS X and now need to add the
capability to play a sound file asking for a 5 digit number, play another
message asking for a 2 digit number, pass these variables to a shell
script, and get the result. I have tried a number of different scenarios
but they are not working. I have read through the wiki, past posts, and
numerous websites.
The sound files are enter-first enter-second
The shell script is CheckNumbers.sh
exten = 2,2,get_data (enter-first,1,5)
exten = 2,3,get_data (enter-second,1,2)
exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber,
secondnumber)
I really appreciate your help!
Jane
Jane,
try this
exten = 2,2,read (firstnumber,enter-first,5)
exten = 2,3,read (secondnumber,enter-second,2)
exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber}
${secondnumber})
I believe it is the syntax that is holding you back.
John M
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Re: [Asterisk-Users] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? Try this in zapata.conf for fun: busydetect=yes busycount=6 Let us know if it makes a difference. I have added this to zapata.conf. Will let you know what happens. can you tell me why this might help or point me to a wiki/google? Going from memory only (which might be less then accurate), the busycount parameter essentially extends zap detect time. The comments in zapata.conf refer to detecting busy tone, but something from past memory says the parameter affects more then just busy tone detection. The default value is 4 but I've been using 6 or at least a year with an x100p followed by a TDM04b, and I don't have the false ring issues. Sure wish I would have kept a diary of config changes over the last two years rather then rely on memory. It would have been helpful more than once. :( Well I added these settings to zapata.conf and am still getting the phantom rings, 2 so far this morning! have been watching ztmonitor and am seeing that that rx audio level is showing a constant ###* with an rx gain setting of -7.5 in zapata.conf. If i set gain much less it gets hard to hear voice from callers. With gain at 0.0 i get * with some peaks above this. Is this normal? John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? Try this in zapata.conf for fun: busydetect=yes busycount=6 Let us know if it makes a difference. I have added this to zapata.conf. Will let you know what happens. can you tell me why this might help or point me to a wiki/google? Going from memory only (which might be less then accurate), the busycount parameter essentially extends zap detect time. The comments in zapata.conf refer to detecting busy tone, but something from past memory says the parameter affects more then just busy tone detection. The default value is 4 but I've been using 6 or at least a year with an x100p followed by a TDM04b, and I don't have the false ring issues. Sure wish I would have kept a diary of config changes over the last two years rather then rely on memory. It would have been helpful more than once. :( Well I added these settings to zapata.conf and am still getting the phantom rings, 2 so far this morning! have been watching ztmonitor and am seeing that that rx audio level is showing a constant ###* with an rx gain setting of -7.5 in zapata.conf. If i set gain much less it gets hard to hear voice from callers. With gain at 0.0 i get * with some peaks above this. Is this normal? John M After watching ztmonitor i have found that the rx audio goes full scale when i get the phantom rings, same as with an actual call. I think I am proving to myself that the problem is in the pots line? I am going to try and put a meter on the pots line and see if I am getting ring voltage on the phantom calls. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phantom incomming calls from asterisk
About once a day I have noticed a phantom incoming call with a caller ID of [EMAIL PROTECTED]cut off. When I answer the call there is a dial tone and the call is disconnected. Any clues? David Koski David and List, I am having the same problem. I have an * box at my house with 1 zap (pstn on a X100p clone from digit networks) channel and one sip(linksys ATA). I am getting ring on the ATA but there is no call comming in from the pstn. The following is the CLI output when this happens. I know that there is no call on the pstn because i have an emergency phone(frequent power outages) still connected to the PSTN parallel to the * box and it never rings. All the SIP stuff is on an internal lan only. I only call out on PSTN since all I have available here in nowheare land is dial up :-( All work flawlessly except for this one problem. - Starting simple switch on 'Zap/1-1' Jul 8 13:49:23 NOTICE[6150]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, sip/677|35) in new stack -- Called 677 -- SIP/677-55a8 is ringing == Spawn extension (default, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Is there anything for Zap like sip debug? My first guess is that I am getting some sort of blip in ring voltage on the PSTN but have no way to prove this. As a posible logic check I unplugged from PSTN, which put zap into Red alarm of course, and then i get no phantom calls. Is there something in the zap driver that shuts down when in red alarm? Any Ideas? John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pbx_extension_helper: No application 'agi'
On Tuesday June 28 2005 4:56 am, Tom Fielding wrote: Hi all, Sorry for this elementary question (I'm a newbie). I'm trying to write an agi script (test.agi) and run it when I call in. However, I'm getting an error that says application agi isn't being found. I've put test.agi into agi-bin with permissions 755. Do I have to compile agi support into Asterisk, or is it built in? My test.agi script is php, but not using anything fancy (just sending me an email) so I didn't install PHP AGI. Do I have to? Thanks, Tom DEBUG: Connected to Asterisk 1.0.7 currently running on dev1 (pid = 26799) Verbosity is at least 10 -- Executing Goto(SIP/4.68.250.152-08129478, validatenumber|s|1) in new stack -- Goto (validatenumber,s,1) Jun 28 01:01:02 WARNING[26800]: pbx.c:1291 pbx_extension_helper: No application 'agi' for extension (validatenumber, s, 1) == Spawn extension (validatenumber, s, 1) exited non-zero on 'SIP/4.68.250.152-08129478' EXTENSIONS.CONF: [validatenumber] exten = s,1,agi(test.agi) exten = s,2,HangUp exten = s,1,agi(test.agi) should be exten = s,1,agi,test.agi If there any arguments to send the script use exten = s,1,agi,test.agi|args_to_pass John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt
Mr. DiMartino, how about you go to the qmail list and stay there so they can listen to your whining and not us. This is a VERY helpful list. Yes there is the occasional question that goes unanswered, but this is rare. Stop trolling, go away, and grow up. Sometimes is is as important to know what not to do, as it is to know what to do. John M On Monday June 27 2005 4:11 pm, Michael Di Martino wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Monday, June 27, 2005 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] LiveVoip is Bankrupt On Mon, 2005-06-27 at 14:31 -0400, Michael Di Martino wrote: If this list spent at least half the time on helping other asterisk admins as it does on trivial things like LiveVoips bankruptcy it just might be a great list. As it stands now this list is kind of useless. Most request for assistance with asterisk problems go unresolved of unanswered. If you would like to see how a good list is run join the Qmail users list and observe. The bankrupt thread is mostly now about finiding hosting for Daily Asterisk News, which I feel is helping asterisk people, and people whining about this thread. The whining seemed to be from people reading the subject line and not even bothering to notice that the majority of the posts under this subject were about an asterisk specific thing when I saw that. This isnt slashdot we should actually read more than the subjects before commenting. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN (PRI) in the US and Redirect?
Hello, I have read about using redirect on a sip channel to get * to step out of the voice path. Is this possible with ISDN or maybe a US T-1? I would like to have the * box answer a call, do some niffty IVR/AGI stuff, and then redirect that call back to the originating switch to be passed on to a dialed number or straight to a dialed number and then step out of the voice path to free up bandwidth. Is this possible or do I need to get involved in SS7 stuff? I have seen a lot of talk on the wiki about redirect but nothing that I understand as what I need. I have seen the manager API redirect to place 2 calls into a meetme but this is not what I need, also not looking for call deflect. Thank you for any sugestions, John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help me please
Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the Digium's TDM20B card. what is the wire combination/sequence for a successful installation? please define the pin no.s accurately. You advice/support is highly appriciated. Thanking you Kumara Kamura, Simply plug the RJ11 into the RJ45 socket. The tab will center the plug and the pinouts are compatable John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNIS and DID seeking confirmation
Hello all, After much googling I have come to the conclusion that in asterisk land DID(Direct Inward Dial) and DNIS(Dialed Number Identification Service) are used rather interchangeably. If this is an incorrect assumption Please correct me. Based on this assumption if I have everthing set up to land in the [incoming] context and an 800# such as 1-800-123-4567 with 4 digit DNIS I can have an entry in my incoming context exten = _4567, 1, do something this is where the call to my 800 number will land regardless of which trunk the call comes in on. Like wise if I have a DID number 456-7891 with an exten= _7891,1,do something else this will also work. Is this correct or am I way off base? Also what is Asterisk looking for as far as a delimiter or is that in a config file? Something like Seize (Wink) DNIS (Wink) ANI (Wink) Answer or Seize (*) DNIS (*) ANI (*) Answer John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toll Free DIDs
I have several toll free numbers that get forwarded to a single local number assigned to a trunkgroup. I've asked the telco to not forward those toll free numbers but to assign them as DIDs to the trunkgroup, so that I can differentiate via DNID. They said that they can't do that. That toll free numbers must forward. I know that I could have them each forward to different local DIDs assigned to the trunkgroup, but that just doesn't seem necessary. Is the telco correct? Technically they are partly correct. 800 numbers are pointed to a local number. Although, they can pass the 800-XXX- to you IF they choose to. In my experience(limited as it may be) it is easier to have them point to a specific local number assigned to the trunkgroup. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pricing for DS3000P
That was a policy we did not adopt, something about using the word 'unlimited' and then not wanting to fill it with a ton of qualifiers like 'its unlimited unless you actually use then then we will limit you, but if you never use it then ...' :) We termed it unlimited interactive meaning you were more or less at the computer. Lots of people on IRC seem to be there 24 hours a day. Lots of people seem to sleep with their computers too. :-) rant This thread is a bit OT but I can't help respond. I live in an area where my only choice is dial up, Directway, or T-1. The first is $10 to $20 a month , Dway $40 and crappy service, T-1 $500+ too much for my budget. I am on the computer checking e-mail get service packs for Windoze a major portion of the day and yet my provider sees fit to disconnect me after 12 hours weather i have been trying to get that stupid 200 meg SP2 for MS crap or not. It would be my opinion that after a time of INACTIVITY sure disconnect, but if there is actual traffic DO NOT DISCONNECT. /rant John M Unlimited service user ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk
On Thursday May 19 2005 8:38 am, Allan Regenbaum wrote: I have been trying for days to get an outbound connection to broadvoice with no luck ..details below ... I have scoured all postings and seem to get similar responses but none of these seem to help... any help is appreciated .. my [EMAIL PROTECTED] box is sitting as 192.168.1.106 behind a linksys router that feeds to comcast as the provider. trying to make outbound calls from a analog phone extension on a digium baord to broadvoice .. system works fine analog phone to analog trunk , but cant get calls out from analog phone or softphone to broadvoice . asterisk log throws -- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 604 Does not exist anywhere back from 147.135.0.128 == No one is available to answer at this time -- Executing Congestion(Zap/1-1, ) in new stack == Spawn extension (from-internal, 17705229625, 2) exited non-zero on 'Zap/1-1 sip .conf is [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 192.168.1.106; Address to bind to (all addresses on machine) disallow=all allow=gsm allow=ulaw allow=alaw ;context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown context = from-broadvoice externip=69.??.??.?? localnet=192.168.1.0/255.255.255.0 sip_additional.conf shows register=561???:91?:@sip.broadvoice.com/201 *** i have tried various permutations of this [bv] username=5618282155 user=phone type=peer secret=myPassword nat=yes insecure=very host=sip.broadvoice.com fromuser=561?? fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband context=from-broadvoice canreinvite=no authname=561?? [sip.broadvoice.com] username=561 user=561 type=user secret=91??? nat=yes insecure=very host=sip.broadvoice.com fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband context=from-broadvoice canreinvite=no also , per postings on the boards ..i pasted this to extensions.conf ..seems that amp had not created an entry for this exten = _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) exten = _1NXXNXX, 2, congestion() exten = _1NXXNXX, 102, busy() == outbound routing ... i have prefix 1 directing to the BV trunk all (other than general section in sip.conf and the extensions.conf) were setup using amp ..seems amp does not place the entries in extension.conf ... === trunks in amp is a follows sip trunk... outbound caller is is broadvoice max channels is blank no dial rules no dial prefix outgoing settings trunk name is bv peer details are authname=561??? canreinvite=no context=from-broadvoice dtmf=inband dtmfmode=inband fromdomain=sip.broadvoice.com fromuser=561??? host=sip.broadvoice.com insecure=very nat=yes secret=91?? type=peer user=phone username=561??? incoming settings user context sip.broadvoice.com user details canreinvite=no context=from-broadvoice dtmf=inband dtmfmode=inband fromdomain=sip.broadvoice.com host=sip.broadvoice.com insecure=very nat=yes secret=91? type=user user=561? username=561 register string ... 561??:91?:@sip.broadvoice.com/201 = fyi ... this is an [EMAIL PROTECTED] setup my bv number is shown as 561?? my bv fancy password is shown as 91?? i have a outbound rule that says numbers with prefix 1 ..go to sip/bv trunk try setting a /etc/hosts entry for one of their proxy servers( I use 147.135.12.128, 147.135.0.128 is not good) if you ping all their proxies and set the hosts entry to the fastest one this will help. ALSO you should know that there are MAJOR problems with broadvoice. I have had an account with them for 3 months or so and at first all worked great, then the last month or so it has been very bad! As of this morning i am getting no sound in either direction. my asterisk box is getting and answering the call, playing the voice prompts that it should but I can not here them and it does not receive any DTMF. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channelized T-1 (NOT PRI) Voice and IP mixed
Hello All, I have googled and wikied but must not be searching correctly. Assuming the TE110P has same ability as old T100P to use some voice and some data channels, lets say I have a TE110P set to accept voice on 10 channels and pass the other 14 channels as data. Under this scenerio i am guessing that * should still be able to accept VoIP calls on the data channels and still allow internal user to access Internet through data portion of the T? Am I correct or am I talking out of my ... I know that a channel bank would be a better solution, but. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Register two account at Broadvoice with one asterisk box
Hello all, I have asked this question of Broadvoice support and the following is their responce: John, Unfortunately we are not able to fully support asterisk. We refer customers to the Asterisk forums where users are quite well versed and some are affiliated with BroadVoice. The only thing that comes to mind is that you may have to specify different ports for each number. Thank you, BroadVoice Customer Care tried voip-inf.org and not getting responce (down? or just me?) I can call in to and out of * from either number/account that i have. The problem is i would like to answer with different prompts based on which account/number the called dialed but broadvoice sends the call as if it came from whichever account i register second. Executing Answer(SIP/xx1492-d5d8, ) in new stack This is the same regardless off the number i call. I have tried register = user:pass@sip.broadvoice.com:5060 for first line and user2:pass2@sip.broadvoice.com:5061 this does not work for me. Is it possible to register on different ports? relevant sip.conf [broadvoice] type=peer username=xx1405 fromuser=xx1405 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very [broadvoice2] type=peer username=xx1492 fromuser=xx1492 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice2 dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very any help is much appreciated Thank you, John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Register two account at Broadvoice with one asterisk box
I can call in to and out of * from either number/account that i have. The problem is i would like to answer with different prompts based on which account/number the called dialed but broadvoice sends the call as if it came from whichever account i register second. Executing Answer(SIP/xx1492-d5d8, ) in new stack This is the same regardless off the number i call. I have tried register = user:pass@sip.broadvoice.com:5060 for first line and user2:pass2@sip.broadvoice.com:5061 this does not work for me. Is it possible to register on different ports? relevant sip.conf [broadvoice] type=peer username=xx1405 fromuser=xx1405 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very [broadvoice2] type=peer username=xx1492 fromuser=xx1492 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice2 dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very any help is much appreciated Thank you, John M from Trixter and reposted at bottom( for ease of information flow) There are a couple ways to do this. Or shoiuld be anyway. One is by setting the context as you have done. The other is by setting the extension at the end of the register line and doing a goto(someplace,s,1) for one line and goto(someplaceelse,s,1) for the other all from the same context. snip Thank you but... this did not help. the problem is that the calls all come in as if from the same account, whichever registers second. called first number and got: -- Executing Answer(SIP/xx1492-b2c7, ) in new stack which should have been xx1405 called second number and got: -- Executing Answer(SIP/xx1492-2f6a, ) in new stack which is correct and is second in register statement Does anyone know if Broadvoice passes the equivalent of DNIS and is there a way to capture that in * from a VoIP call? John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this normal - Long time to make call - What is your average with your Hardware?
Hardware - Pentium 1.4 Gig - 1 Meg ram - 1 FXO100 Card - Sipura 2000 - Local Network Router SMC -Codec 711 - Asterisk @ home (lastest) On average it take almost 10 - 13 Secs to make an outbound call to a local number. Is this a normal time ? Is there something that can be done to cut this time down? Is the FXO100 the problem ? ie. Modem card ? May not be relevant but there is a dial plan setting in the sipura that waits until a pattern is matched and waits until time out if none are matched before sending digits(dtmf) on to *. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Concurrent calls: best provider?
On Monday April 04 2005 3:58 pm, Kevin Kiely wrote: T1 PRI This brings up the question. What is the best service for concurrent calls? In the case where I have a small business I might have 10-15 people needing to call out and they could all be on at the same time. -Scott Even with a T-1 you still need some one to provide termination that will allow more than one call at a time on that account or multiple accounts with the same or different providers. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Concurrent calls: best provider?
On Monday April 04 2005 5:14 pm, Brian McSpadden top posted: I believe Kevin is meaning to buy a T-1 PRI from someone, like a phone company of a CLEC. On Apr 4, 2005 3:40 PM, John Millican [EMAIL PROTECTED] wrote: On Monday April 04 2005 3:58 pm, Kevin Kiely wrote: T1 PRI This brings up the question. What is the best service for concurrent calls? In the case where I have a small business I might have 10-15 people needing to call out and they could all be on at the same time. -Scott Even with a T-1 you still need some one to provide termination that will allow more than one call at a time on that account or multiple accounts with the same or different providers. John M Well I canget T-1 from a local provider (since i live in BFE it is much cheaper than att, verizon,...) but they do not provide termination. so just wanted to clarify this for the op. john M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of thisemailforum??
On Thursday March 31 2005 11:28 pm, Tim Bass wrote: This list certainly needs a moderator. Mr. Bass, I like this list,. It has a GREAT DEAL of usefull information that has helped me become proficient in the use of asterisk, and has provided some comic relief during the day. If you don't like it leave. Just like television, If you don't like the show turn the channel. It is called maturity and doesn't need a nanny or a moderator. John Millican ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working
On Tuesday March 08 2005 2:58 pm, James Taylor wrote:
Ok, used your sip.conf inbound works. Outbound gets:
SIP/2.0 604 Does not exist anywhere
Any ideas?
James
On Tue, 8 Mar 2005 14:18:11 -0500, Marios Andreou [EMAIL PROTECTED]
wrote:
Yes it is working just fine for me with the same sip.conf that you have.
??
Except the permit=sip.broadvoice.com
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
Also what is your extensions.conf ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not
working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis [EMAIL PROTECTED]
snip
I have a same sip.conf and out and in are working well. I have
sip.broadvoice.com mapped to proxy.lax.broadvoice.com in my hosts file. this
is nice for me as i can use sip.broadvoice.com in all .conf and if i need to
change the proxy i do so in the hosts file. I do not use ${EXTEN:1} in my
outbound dial and i always dial 10 digits.
John Millican
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Re: [Asterisk-Users] Dial out through Broadvoice
On Saturday February 26 2005 4:45 pm, John Millican wrote:
On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was calling
to test knowing it would not be busy and would not bother anyone.
Make sure you have your iax set up right in the Iax.conf and your
outbaound registering string going back out.
I have mine set up that I dial 6 to get out on my broadvoice line and 9
to get out on my voice pulse line.
I am not using IAX at all. Did not think broadvoice supported it, am I
wrong?
More Comments at BOTTOM
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily Not Available back from
147.135.16.128
-- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack
== Spawn extension (outgoing, 16037862111, 102) exited non-zero on
'SIP/147.135.0.129-0815bc60'
Is this as simple as it seems? Broadvoice is circut busy? Can any
one think
of any other reason I might get this message? Or do I just need to
call
BroadVoice and complain? I have tried two different proxy's (ip's in
/etc/hosts) and get the same error.
in extensions.conf:
[outgoing]
exten = _1NXXNXX, 1, dial(SIP/${EXTEN}
@proxy.bos.broadvoice.com,30) ;
exten = _1NXXNXX, 2, congestion() ; No answer, nothing
exten = _1NXXNXX, 102, busy() ; Busy
in sip.conf:
[general]
context=default ; Default context for incoming
calls
port=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=192.168.123.100 ; IP address to bind to
(0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables
the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
register =
[EMAIL PROTECTED]:XX:[EMAIL PROTECTED]
[broadvoice1]
type=friend
username=603XXX
fromuser=603XXX
secret=XX
host=proxy.bos.broadvoice.com
fromdomain=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes
[bv-in-1]
type=friend
host=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
Try adding this line to sip:
insecure=very
Added insecure=very and same message
see if that helps. if not, try a standard registration string instead
of the one broadvoice tells you to use.
Also - make sure you're using the password they sent you in an email -
not the one you used when you signed up on their website.
Registration seems to work and shows as registered when i run sip show
registry.
I have been on the support line with broadvoice several times now and still no
resolution, so I am askking help again, please. Below is sip show registry
and a sip debug. Does any one have any sugestions? I followed the
instrution at http://edvina.net/broadvoice/ along with others and sytill no
luck on outbound calls. in the sip debug it is showing the internal ip in
the callid field. I have externip= external ip in sip .conf am i missing
something else?
linux*CLI sip show registry
HostUsername Refresh State
sip.broadvoice.com:5060 [EMAIL PROTECTED]15 Registered
linux*CLI sip debug
SIP Debugging Enabled
Feb 28 11:08:34 NOTICE[5214]: chan_sip.c:3999 sip_reregister:--
Re-registration for [EMAIL PROTECTED]@sip.broadvoice.com
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.123.100:5060;branch=z9hG4bK01ee8f5f
From: sip:[EMAIL PROTECTED];tag=as2999a955
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-Length: 0
(no NAT) to 147.135.8.128:5060
linux*CLI
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.123.100:5060;received=69.160.185.49;branch=z9hG4bK01ee8f5f;rport=63364
From: sip:[EMAIL PROTECTED];tag=as2999a955
To: sip:[EMAIL PROTECTED];tag=SD30scc99-
Call-ID: [EMAIL PROTECTED]
CSeq: 104 REGISTER
Contact: sip:[EMAIL PROTECTED];expires=20
Content-Length: 0
8 headers, 0 lines
Feb 28 11:08:34 NOTICE[5214]: chan_sip.c:6779 handle_response: Outbound
Registration: Expiry for sip.broadvoice.com is 20 sec
Re: [Asterisk-Users] Dial out through Broadvoice
missed somethins but i do not know what/ I greatly apreciate all the help so far. John Millican ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial out through Broadvoice
On Monday February 28 2005 6:16 pm, Roger Hanson wrote: - Original Message - From: Gabriel Gunderson [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 28, 2005 4:49 PM Subject: Re: [Asterisk-Users] Dial out through Broadvoice Am i not providing some helpfull info? If not tell me what i am missing and i will get it. I am sure I have missed somethins but i do not know what/ I greatly apreciate all the help so far. John Millican The service might just be down. I was up and working just fine and a few hours ago that changed. I'll just wait and see if things get better. BV is kinda like that - a little flaky at times. I just wonder if it has anything to do with asterisk or if all of their customers get the same. Gabe ___ I haven't noticed any downtime in quite a long time - but maybe just missed it. I have been using my broadvoice trunk fairly often with no hint of problems. I've even been considering dumping my PSTN phone lines and going strictly VoIP through Broadvoice at home. But more testing needs to be done first. So, what exactly is happening again? You can rx calls but not tx calls over Broadvoice? Correct? Can you rx calls over any other VoIP provider or PSTN? Could you post your current configs again? I was unable to tx could rx all day no problem i was getting an error: SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 480 Temporarily Not Available back from 147.135.16.128 -- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy == Everyone is busy/congested at this time -- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack == Spawn extension (outgoing, 16037862111, 102) exited non-zero on 'SIP/147.135.0.129-0815bc60' was the origanal error. I tried all the other proxy's and I then got a does not exist anywhere error. I finally got it working about an hour ago. here is the working sip.conf [broadvoice] type=peer // changed to peer from friend. very simple once i knew username=603xxx fromuser=603xxx secret=bvpassword host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very This is working now, althouh i get chopped ringback , but once the call path is set the audio is good. Thanks again for all the help John Millican ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial out through Broadvoice
snip So, what exactly is happening again? You can rx calls but not tx calls over Broadvoice? Correct? Can you rx calls over any other VoIP provider or PSTN? Could you post your current configs again? I was unable to tx could rx all day no problem i was getting an error: SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 480 Temporarily Not Available back from 147.135.16.128 -- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy == Everyone is busy/congested at this time -- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack == Spawn extension (outgoing, 16037862111, 102) exited non-zero on 'SIP/147.135.0.129-0815bc60' was the origanal error. I tried all the other proxy's and I then got a does not exist anywhere error. I finally got it working about an hour ago. here is the working sip.conf [broadvoice] type=peer // changed to peer from friend. very simple once i knew username=603xxx fromuser=603xxx secret=bvpassword host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very This is working now, althouh i get chopped ringback , but once the call path is set the audio is good. Thanks again for all the help John Millican just a ps in /etc/hosts i have sip.broadvoice.com mapped to the ip of proxy.lax.broadvoice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial out through Broadvoice
Hello all, When I call the Broadvoice number all is good. When I try to call out through DISA on my broadvoice line i get the following: Executing Dial(SIP/147.135.0.129-0815bc60, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 480 Temporarily Not Available back from 147.135.16.128 -- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy == Everyone is busy/congested at this time -- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack == Spawn extension (outgoing, 16037862111, 102) exited non-zero on 'SIP/147.135.0.129-0815bc60' Is this as simple as it seems? Broadvoice is circut busy? Can any one think of any other reason I might get this message? Or do I just need to call BroadVoice and complain? I have tried two different proxy's (ip's in /etc/hosts) and get the same error. in extensions.conf: [outgoing] exten = _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) ; exten = _1NXXNXX, 2, congestion() ; No answer, nothing exten = _1NXXNXX, 102, busy() ; Busy in sip.conf: [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=192.168.123.100; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet register = [EMAIL PROTECTED]:XX:[EMAIL PROTECTED] [broadvoice1] type=friend username=603XXX fromuser=603XXX secret=XX host=proxy.bos.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes [bv-in-1] type=friend host=sip.broadvoice.com context=broadvoice dtmfmode=inband canreinvite=no nat=yes Thank you for any help. John Millican ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial out through Broadvoice
On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was calling to
test knowing it would not be busy and would not bother anyone.
Make sure you have your iax set up right in the Iax.conf and your outbaound
registering string going back out.
I have mine set up that I dial 6 to get out on my broadvoice line and 9 to
get out on my voice pulse line.
I am not using IAX at all. Did not think broadvoice supported it, am I wrong?
More Comments at BOTTOM
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily Not Available back from
147.135.16.128
-- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack
== Spawn extension (outgoing, 16037862111, 102) exited non-zero on
'SIP/147.135.0.129-0815bc60'
Is this as simple as it seems? Broadvoice is circut busy? Can any
one think
of any other reason I might get this message? Or do I just need to
call
BroadVoice and complain? I have tried two different proxy's (ip's in
/etc/hosts) and get the same error.
in extensions.conf:
[outgoing]
exten = _1NXXNXX, 1, dial(SIP/${EXTEN}
@proxy.bos.broadvoice.com,30) ;
exten = _1NXXNXX, 2, congestion() ; No answer, nothing
exten = _1NXXNXX, 102, busy() ; Busy
in sip.conf:
[general]
context=default ; Default context for incoming
calls
port=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=192.168.123.100 ; IP address to bind to
(0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables
the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
register =
[EMAIL PROTECTED]:XX:[EMAIL PROTECTED]
[broadvoice1]
type=friend
username=603XXX
fromuser=603XXX
secret=XX
host=proxy.bos.broadvoice.com
fromdomain=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes
[bv-in-1]
type=friend
host=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
Try adding this line to sip:
insecure=very
Added insecure=very and same message
see if that helps. if not, try a standard registration string instead
of the one broadvoice tells you to use.
Also - make sure you're using the password they sent you in an email -
not the one you used when you signed up on their website.
Registration seems to work and shows as registered when i run sip show
registry.
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Re: [Asterisk-Users] cepstral integration with * using AGI?
On Monday January 24 2005 3:29 pm, John Middleton wrote:
Hi, I've looked at the Wiki for this, have seen the Swift.agi
details, but has anyone got a current script for Cepstral and an
example of integraton in * please?
I'm a * and linux newbie, so please be gentle ;-)
Thanks
John
I just put swift.agi in agi-bin and used a c++ script to do a db
look-up in postgres for the information that i wanted read to the
user, i.e user name and other info based on caller id number. I pass
calleridnum to the c++ script and then use SETVAR in the script to
get the info back and read it to the user. recfound is set to 1 if
any record in the db.
Extensions .conf look somewhat like this:
[answerMain]
exten = s,1,Ringing() ; Send Ring tone to caller
exten = s,2,Wait,5 ; Wait a 5 seconds to get a
ring or two
exten = s,3,Answer ; Answer the line
exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,5,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten = s,6,PrivacyManager
exten = s,7,agi,c++script.cpp|${CALLERIDNUM}; //does db lookup
exten = s,8,GoToIf($[${RECFOUND} 0]?9:17); // if found a record
exten = s,9,agi,swift.agi|Welcome to the Reservation System. We will
be placing a reservation for ${varname}.;
exten = s,10,read(foo,static recording,1); //wait for user input of 1
digit
I us QT for writting the script but it is just a simple C++ script,
subject for different mail list.
John M
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Re: [Asterisk-Users] cepstral integration with * using AGI? -sent last responce to soon stupid me
On Monday January 24 2005 3:29 pm, John Middleton wrote:
Hi, I've looked at the Wiki for this, have seen the Swift.agi
details, but has anyone got a current script for Cepstral and an
example of integraton in * please?
I'm a * and linux newbie, so please be gentle ;-)
Thanks
John
I just put swift.agi in agi-bin and used a c++ script to do a db
look-up in postgres for the information that i wanted read to the
user, i.e user name and other info based on caller id number. I pass
calleridnum to the c++ script and then use SETVAR in the script to
get the info back and read it to the user. recfound is set to 1 if
any record in the db.
Extensions .conf look somewhat like this:
[answerMain]
exten = s,1,Ringing() ; Send Ring tone to caller
exten = s,2,Wait,5 ; Wait a 5 seconds to
get a ring or two
exten = s,3,Answer ; Answer the line
exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5
seconds
exten = s,5,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten = s,6,PrivacyManager
exten = s,7,agi,c++script.cpp|${CALLERIDNUM}; //does db lookup
exten = s,8,GoToIf($[${RECFOUND} 0]?9:17); // if found a record
exten = s,9,agi,swift.agi|Welcome to the Reservation System. We will
be placing a reservation for ${varname}.;
exten = s,10,read(foo,static recording,1); //wait for user input of 1
digit
I us QT for writting the script but it is just a simple C++ script,
subject for different mail list. but here is an example.
#include qsqldatabase.h
#include qdatatable.h
#include qsqlcursor.h
#include qsqlquery.h
#include qstring.h
#include stdio.h
#include qapplication.h
#include iostream
#include qregexp.h
#include qdatetime.h
#include qprocess.h
using namespace std;
#define DRIVER QPSQL7/* PostgreSQL Driver*/
#define DATABASE DBNAME /* the name of the database */
#define USER jmillican /* user name with
appropriate rights */
#define PASSWORD**/* password for USER */
#define HOST 127.0.0.1 /*host on which the database
is running */
QSqlDatabase * db ;
int main( int argc, char **argv)
{
bool useGUI = false;
QApplication a( argc, argv, useGUI);
setlinebuf(stdout);
setlinebuf(stderr);
QString callid;
QString astvar;
callid = a.argv()[1]; //get caller id from asterisk
QSqlDatabase * db = QSqlDatabase::addDatabase( DRIVER );
db-setDatabaseName( DATABASE );
db-setUserName( USER );
db-setPassword( PASSWORD );
db-setHostName( HOST );
if (!db-open())
{
fputs(db not open \n,stderr);
}
// get cust_id and stuff from callid
QSqlQuery query;
query.prepare(select SQL QUERY HERE);
query.bindValue(:phone,callid);
query.exec();
query.next();
QString strCustId = query.value(0).toString();
QString strsomeId = query.value(1).toString();
QString strsomeName;
QVariant vSize = query.size();
QString strSize = vSize.toString();
strsomeName = query.value(2).toString();
strsomeName = strBoatName.replace( ,,);
if (query.size() = 1) //send all info back to asterisk
{
fprintf(stdout,EXEC SETVAR CUSTID= + strCustId + \n);
fprintf(stdout,EXEC SETVAR BOATNAME= + strsomeName + \n);
fprintf(stdout,EXEC SETVAR BOATID= + strsomeId + \n);
fprintf(stdout,EXEC SETVAR RECFOUND= + strSize + \n);
}
else
{
fprintf(stdout,EXEC SETVAR RECFOUND=0 \n);// if no record
found
}
db-close();
return 0;
If any one sees this as a bad example please say so with comment on
how to make it better.
John M
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Re: [Asterisk-Users] callers who don't press any keys
Warren Burstein wrote: I've noticed that some callers listen to our main menu and don't press any keys. snip Remember Rotary Phones? They are still in use in some homes/areas John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel init script
-- Original message -- From: WipeOut [EMAIL PROTECTED] I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having an issue with the zaptel init script.. If I run.. #modprobe zaptel #modprobe wcfxo #modprobe wcfxs .. from a command line it load and appears to be working fine.. If I try and use the init script I get errors about ZT_CHANCONFIG and the modules don't see to laod up.. Anyone got any pointers? I am running Fedora Core 2 with all the updates and I have an X100P and a TDM400P with a single FXS module.. Later.. I belive I have seen on the list where wcfxs has been changed to wctdm this may be your problem? John Millican --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 11/1/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Advice on OS Choice
Yes I am lazily top posting. This thread is getting ridiculous (don't waste your time flaming). Freedom is being able to choose if I want to use GPL or BSD or my own stupid license agreement. To give my work away for no cost or to charge exobanant prices for it. We are not all ever going to agree one is better than the other, there will always be some disagreement. That is the beauty of freedom, we can disagree. If I like one better than the other I will use it and you are free to think that I am stupid and use the other. That is freedom! I don't have to agree with you, you don't have to agree with me and we can both say so without fear of governmental reprisal. Now lets get back to talking about the wonderful software that this list is about, Please. I read this to learn about Asterisk, not GPL or BSD. Thank you for your time, John Millican -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Kohlsmith Sent: Friday, October 15, 2004 4:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Advice on OS Choice On Friday 15 October 2004 16:22, Michael Giagnocavo wrote: problem lies in the policy for upgrading or installing software on life-critical machines not being followed. I agree with that. But, what's going to be held up in court? As a lawyer for a medical equipment corp, which route are you going to take to be safe? As a medical equipment corp system designer (I do this for a living, although not for medical) I'd make damn sure the software couldn't be updated without the correct access codes being in place, including hardware interlocks with physical keys. It's not hard to make firmware loaders require this kind of stuff. Imagine a toaster that ships with a booklet that shows the schematics and shows people how to rebuild the toaster. Then some person (either a 9-yr-old or an experienced electrician) uses the instructions, and fries themselves. Or the next person who uses the toaster starts a fire. When it gets to court, you can bet that the lawyers are going to try to blame the company for making it easier to modify the toaster. Even though it's utterly silly, that's how the US legal system works. No one is responsible for their own mistakes. This used to be the way it was. The Amiga computers all came with full schematics. Radios and televisions had easily obtainable service manuals. Radio Shack actually had a decent parts inventory. Hell IIRC certain versions of DOS (CP/M?) had full source listings! *sigh* good old days... -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 10/8/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 10/8/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys PAP2-NA
Eric, I was told by Bottom Line Tech that Linksys told them to pull all units and stop all shipments unless there customer could prove they were and ISP, which i am not so i can not, so no [EMAIL PROTECTED] for ME :-( John Millican -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Merkel Sent: Wednesday, September 22, 2004 10:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Linksys PAP2-NA I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that they heard from Cisco that these units were not due out until Dec. Did Cisco/Linksys pull these units off the shelves? -- Eric Merkel MetaLINK Technologies, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys PAP2-NA
Here is the contact info For Bottom Line Tech Bottom Line Telecommunications www.shopblt.com 457 Route 164 Preston, CT 06365-8111 (860) 886-1011 / (561) 791-3308 John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary Carr Sent: Wednesday, September 22, 2004 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys PAP2-NA You have some contact info for Bottom Line Tech? We are a ISP/CLEC and want to order some of these. Gary Eric, I was told by Bottom Line Tech that Linksys told them to pull all units and stop all shipments unless there customer could prove they were and ISP, which i am not so i can not, so no [EMAIL PROTECTED] for ME :-( John Millican -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Merkel Sent: Wednesday, September 22, 2004 10:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Linksys PAP2-NA I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that they heard from Cisco that these units were not due out until Dec. Did Cisco/Linksys pull these units off the shelves? -- Eric Merkel MetaLINK Technologies, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys PAP2-NA
Well just did a search on bottom line and they do not have the PAP2-NA listed anymore. They may still have them in stock if you call them though. Sorry John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Millican Sent: Wednesday, September 22, 2004 3:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Linksys PAP2-NA Here is the contact info For Bottom Line Tech Bottom Line Telecommunications www.shopblt.com 457 Route 164 Preston, CT 06365-8111 (860) 886-1011 / (561) 791-3308 John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary Carr Sent: Wednesday, September 22, 2004 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys PAP2-NA You have some contact info for Bottom Line Tech? We are a ISP/CLEC and want to order some of these. Gary Eric, I was told by Bottom Line Tech that Linksys told them to pull all units and stop all shipments unless there customer could prove they were and ISP, which i am not so i can not, so no [EMAIL PROTECTED] for ME :-( John Millican -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Merkel Sent: Wednesday, September 22, 2004 10:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Linksys PAP2-NA I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that they heard from Cisco that these units were not due out until Dec. Did Cisco/Linksys pull these units off the shelves? -- Eric Merkel MetaLINK Technologies, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys PAP2-NA
This is the guy that i talked to and he seemed helpfull David Durel Bottom Line Telecommunications http://www.shopblt.com/ [EMAIL PROTECTED] Voice / FAX: (860) 886-1011 Monday - Thursday, 9:00 - 6:00 Eastern Time as i said before i searched the site and the PAP2-NA is no longer listed. May be a flood of calls will prompt them to b*%# at linksys. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bartosz Jozwiak Sent: Wednesday, September 22, 2004 3:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys PAP2-NA I would love to have contact info for Bottom Line Tech also. Then we do not have to go with all the trouble getting to them. - Original Message - From: Gary Carr [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 22, 2004 3:59 PM Subject: Re: [Asterisk-Users] Linksys PAP2-NA You have some contact info for Bottom Line Tech? We are a ISP/CLEC and want to order some of these. Gary Eric, I was told by Bottom Line Tech that Linksys told them to pull all units and stop all shipments unless there customer could prove they were and ISP, which i am not so i can not, so no [EMAIL PROTECTED] for ME :-( John Millican -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Merkel Sent: Wednesday, September 22, 2004 10:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Linksys PAP2-NA I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that they heard from Cisco that these units were not due out until Dec. Did Cisco/Linksys pull these units off the shelves? -- Eric Merkel MetaLINK Technologies, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sanity Check --Zapras With T-1
Hello All, I am planning on setting up an * server for a customer and was hoping to get a sanity check on my Plan. What I am trying to accomplish is a * voice and 16 data channel T-1 connection (ESF/B8ZS). I am planning on using a 2.8 ghz P4, 1gig ram, on an Abit AS* Mobo, probably 3Com 10/100nic, T100p, 2 X TDm4xxp(4 ports FXS). Does this sound like a reasonable configuration. Call levels will be light with generally not more that two or three calls at any given time. I could talk customer into using only 1 TDM4xxp since he does not need all 8 FXS. Knowing him he will first want all 8 but... I am then going to pull 16 (1024kbps) channel off of T-1 for data through ZapRas. Am I crazy or will this work. From what i have read it should work very well at least if i go with the X100p and one TDM card, not sure about 2 TDM and X100p in the same machine. Thank you in advance for any help. John Millican --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Background() command
Use Read instead of background CLI show application Read -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Penrod Sent: Friday, September 17, 2004 6:53 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Background() command Folks, Apologies ahead of time if this has already been asked (read the list for the last month looking for something similar). I have been trying to get the Background command to work with no joy yet. Here is what I am trying to do: 1. Answer the call. 2. Play the message in the background, while waiting on DTMF from user. 3. If I get a 1, then interrupt the message and dial the phone number listed at extension 1, otherwise play invalid extension and cycle back. Here is what I have so far: * note: phone numbers changed to generic [message] exten = s,1,Answer() exten = s,2,Wait(2);Pause to let the user end catch up with the connection exten = s,3,Background(demo-congrats) exten = s,4,Goto(3) exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup() exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,3) exten = 1,1,Playback(transfer) exten = 1,2,Dial(SIP/4805551212/20) [incoming] exten = 8665551212,1,Goto(message,s,1) What I get is: 1. The number is answered and the demo-congrats file plays. 2. No matter what button I press on the phone, the file continues to play and recycle when it's done. Question(s): 1. Is there a proper way to configure this? 2. Am I missing a configuration step somewhere in the one of the conf files. Thanks in advance, ...Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.756 / Virus Database: 506 - Release Date: 9/8/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.756 / Virus Database: 506 - Release Date: 9/8/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: how to collect user entered digits
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen R. Besch Sent: Wednesday, August 25, 2004 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: how to collect user entered digits Ryan Courtnage wrote: Had not seen anything on Read anywhere else, must have been looking in all the wrong places. this is a simple solution to my problem. FYI - Read() is described on the wiki: http://voip-info.org/wiki-Asterisk+cmd+Read ___ Even better, bookmark this link: http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+comma nds Thank You all Very Much The above links have been very helpful. John --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.742 / Virus Database: 495 - Release Date: 8/19/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to collect user entered digits
On Fri, Aug 20, 2004 at 01:19:54PM -0400, John Millican said: I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use Get Data filename, timeout, maxdigits in the agi script. In * console I get message saying playing filename but it exits as soon as it starts. Could I collect the digits some how before going to script and then send as an arg maybe? Check out the source to the privacy manager for a simple example... apps/app_privacy.c Well Mr. Reed, and all others on list who might help, I guess I am not as smart as I hoped I was:-) I looked at the apps_privacy file. I thought I could write something similar and compile then call from dial plan. What I get is a lot of errors about unreferenced ast_app_streamfile, ast_app_getdata, ... among others. I used all the same includes in Marks script but still no go. I tried to find where these references came from in the apps_privacy and could not find them. I typically code in C++ with QT 3.3 and would like to be able to use the DB drivers that I have in QT for PostgreSQL since I will be querying an app on a separate machine for customer info based on callid and a date in the form of digits dialed by the caller. Any help in this direction would be greatly appreciated. My original plan was an AGI script, is this a bad idea? Call levels will most likly not be very high. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.742 / Virus Database: 495 - Release Date: 8/19/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to collect user entered digits
Thanks all for your patience. I found my answer in a post on a totally unrelated question (I new if I kept reading all posts ...) saw this in a post for call forwarding. exten = *72,1,Answer exten = *72,2,Wait(1) exten = *72,3,BackGround(allison7/please-enter-your) exten = *72,4,Playback(extension) exten = *72,5,Playback(then-press-pound) exten = *72,6,Playback(beep) exten = *72,7,Read(fromext) Had not seen anything on Read anywhere else, must have been looking in all the wrong places. this is a simple solution to my problem. I have been reading VOIP-info.org wiki, asterisk handbook, google searches. Can anyone point me to some other places that I can search through? John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Millican Sent: Tuesday, August 24, 2004 4:13 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] how to collect user entered digits On Fri, Aug 20, 2004 at 01:19:54PM -0400, John Millican said: I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use Get Data filename, timeout, maxdigits in the agi script. In * console I get message saying playing filename but it exits as soon as it starts. Could I collect the digits some how before going to script and then send as an arg maybe? Check out the source to the privacy manager for a simple example... apps/app_privacy.c Well Mr. Reed, and all others on list who might help, I guess I am not as smart as I hoped I was:-) I looked at the apps_privacy file. I thought I could write something similar and compile then call from dial plan. What I get is a lot of errors about unreferenced ast_app_streamfile, ast_app_getdata, ... among others. I used all the same includes in Marks script but still no go. I tried to find where these references came from in the apps_privacy and could not find them. I typically code in C++ with QT 3.3 and would like to be able to use the DB drivers that I have in QT for PostgreSQL since I will be querying an app on a separate machine for customer info based on callid and a date in the form of digits dialed by the caller. Any help in this direction would be greatly appreciated. My original plan was an AGI script, is this a bad idea? Call levels will most likly not be very high. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.742 / Virus Database: 495 - Release Date: 8/19/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.742 / Virus Database: 495 - Release Date: 8/19/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.742 / Virus Database: 495 - Release Date: 8/19/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to collect user entered digits
Hello all,
I have been searching thru all docs that I can find on wiki and such but can
not get an answer. I am trying to collect a date from user input in the
form of digits dialed from the phone to use in an agi script to do a
database look up. I have tried to use Get Data filename, timeout,
maxdigits in the agi script. In * console I get message saying playing
filename but it exits as soon as it starts. Could I collect the digits some
how before going to script and then send as an arg maybe?
in extensions.conf:
exten = 657, 1, Ringing
exten = 657, 2, wait(5);
exten = 657, 3, BackGround(1)
exten = 657, 4, agi,callid.c|${CALLERIDNUM}
the important part of the agi script:
printf(GET DATA abandon-all-hope, 5000, 6 \\\n);
the message in * console:
Executing AGI(Zap/1-1, callid.c|11) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/callid.c
-- Playing 'abandon-all-hope' (language 'en')
== Spawn extension (default, 657, 4) exited non-zero on 'Zap/1-1'
agi_request: callid.c
-- Hungup 'Zap/1-1'
What am I doing Wrong?
Thank you very much for any help
John Millican (a newbie obviously)
---
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RE: [Asterisk-Users] how to collect user entered digits
Thanks will do that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Walt Reed Sent: Friday, August 20, 2004 2:13 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] how to collect user entered digits On Fri, Aug 20, 2004 at 01:19:54PM -0400, John Millican said: I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use Get Data filename, timeout, maxdigits in the agi script. In * console I get message saying playing filename but it exits as soon as it starts. Could I collect the digits some how before going to script and then send as an arg maybe? Check out the source to the privacy manager for a simple example... apps/app_privacy.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.740 / Virus Database: 494 - Release Date: 8/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.740 / Virus Database: 494 - Release Date: 8/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
