[Asterisk-Users] replacing step-by-step giving echo
hi there, We just encountered the following.. a customer has a tradifional PBX that runs next to asterisk. Both PBX's have their own E1 line. Now 'some' numbers are forwarded from the traditional PBX to the new asterisk server. (both have different DID numbers assigned) When those numbers are called, basically the call arrives on the traditional PBX, gets forwarded to the new number, meaning it goes out the same E1 again, to the new E1 (different telecom operators). Those calls are encountering echo most of the time. When dialing into the E1 on the asterisk server directly, all goes well. (no echo) Should I look into asterisk, the traditional PBX, or even the telecom operator? cheers.. ___ --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] call queue problems
thanks for your help, I really appreciate itOn 4/25/06, Kevin Smith [EMAIL PROTECTED] wrote:
Yes there is. QUEUE_MEMBER_LIST(queuename)This should return you a list of comman-separated list of the members in
a queue. After that you would need to format it (if needed) so asteriskcan read it back to you. Of course then you can make some logicdecesions on whether you want to remove the memeber from the queue, etc.
Also you may find this page helpful for things you are looking forhttp://www.voip-info.org/wiki/view/Asterisk+functionsKevinDumpolid Exeplish wrote:
Thanks Kevin, the tip worked like a charm. However, there are newer issues now! Is there any way of knowing which users are looed in? sometimes, customer support users forget to login B4 they shutdown their computers (we use
soft phones) and presistentmembers=yes is set in queues.conf so the users are not logged off automatically . I have an extension on which I dial to get the count of loged in users. Is there a way to find out
which extensions are currently logged in?? Thanks agai On 4/24/06, *Kevin Smith* [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: Hi, What I would suggest doing, since we have a similar setup (where our 24 support contracts can enter a pin number to be routed to an on call
tech) is I use the function QUEUEAGENTCOUNT(queue name). Since you said that the calls should only be routed after the last support person logs out, just do a test to see if there is anyone logged in the queue, if
not, send them to the NOC. example: exten = s,1,gotoif,$[${QUEUEAGENTCOUNT(124)} 0]?YES:NO exten = s,n(YES),queue(124) ;Since there are more then 0 people
in your queue exten = s,n(NO),queue(123) ; If there less then or equal to 0 You also can run other tests and use logic and's and or's to make the tests more complex.
Hopefully this will help, Kevin
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[Asterisk-Users] Extreme delay before * processes call files
Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call file just keeps sitting in the /var/spool/asterisk/outgoing directory and it seems that * is doing nothing with it?? Only after 10-30 seconds sometimes even much longer the call file is picked up. There is no message on the * console about a call file being present. Does anyone have a clue why asterisk fails to pick up call files within a reasonable amount of time? The load on the box is 0.05 at most. Thanks!! Remco ___ --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] treating an incoming call as a local extension
Hi, Check the DISA command. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jnuoiqweahf kajhdsff Sent: Thursday, April 27, 2006 12:21 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] treating an incoming call as a local extension I have [EMAIL PROTECTED] running on one machine, with X-lite running on another machine on my local network, with X-lite logged in to asterisk as extension 200. From X-lite, I can dial *97 to hear voice mail for extension 200, dial 201 to call extension 201, etc. I need to be able to accept an incoming call over the voip trunk which I have set up, and have asterisk treat that call as extension 202, so that e.g. I can dial in to asterisk from an external voip line and then as soon as asterisk answers the line, I can enter enter a password and then have the call treated as extension 202 and enter *97 to access the voicemail for 202, or enter 200 to locally call extension 200, etc. How can I do this? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Pattern matching problem
So sorry, the correct version is 1.2.6 :-) kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Thursday, April 27, 2006 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: Pattern matching problem My * version is 2.1.6. ... Did I miss something? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hi...Please help me
Hi Chandra, I am also new to Asterisk and I have only just started installing a test system but I probably can help clarify one or two things. I think asterisk "clients" are phones not PCs unless you use"soft phones" which is software onthe PC(somewhat like Skype) that you use to make and answer phone calls. So you might not need to install anything on your PCs if you will use IP phones or ATAs as mentioned by Gonzalo. The hardware you need depends on what you require your asterisk to do. If you will be making only IP calls using IP phones, then you only need asterisk running on your server with no extra hardware. But if you need to connect with analog/digital phone equipment, then you need extra hardware on the server. You do not physically connect your VOIP phone to the asterisk server. You connect it to the network that has the server through a normal network point and configure it to find the server. You probably ought to take Gonzalo's advice and head over to: http://www.voip-info.org/wiki-Asteriskand do some reading before you even start as it will help you fit many pieces of the asterisk "puzzle" together. It helped me get started. Then you probably will have fewer questions that list members will answer more readily :) Regards Wafula From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy BoySent: 26 April 2006 14:56To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Hi...Please help me Hi,Thank you for your response. Basically, I follow "O Reilly AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install Asterisk in server. But, they have not mentioned What I have to install in client PC's? What hardware I need? How can I take decission to buy extra hardware (like Zaptel products) OR no need of buying extra hardware? ( I will be using Asterisk for 70 PC's and a server) Is it sufficient to buy hardware for server only OR for client PC's also? How can I connect my VoIP phone to server? How can I connect hardware to server? How can I connect PSTN line to server PC?Please guide me to complete this task. Waiting for your response. Thank you.Regards,Chandra.Gonzalo Servat [EMAIL PROTECTED] wrote: On 4/24/06, Crazy Boy <[EMAIL PROTECTED]>wrote: Hi Friends,[..snip..] --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) --- -- --- -- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using "Asterisk" software? If It can be implemented using "Asterisk" software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware?[..snip..]It can be done with Asterisk. For the server side, you would need toinstall Asterisk on your Fedora 5 box, Zaptel and lots of Wikireading.I don't recommend using softphones for your employee PCs. It lookslike an attractive solution at first (from a cost perspective) but inreality it's not very practical (at least that was my experience).Buying 5 x 2 port ATAs will cost you around $300-$350 which is notreally expensive considering the kind of powerful PBX you will have atyour disposal. I would have suggested some Digium hardware for the FXS(extensions) but I think it will be a lot more expensive (for 10extensions) than the ATAs solution. You could also look into a channelbank, but again it will be more expensive than the 5 ATAs. As for theFXO (incoming/outgoing PSTN) I recommend buying Digium hardware(TDM400P).Hope this helps, and good luck!Regards,Gonzalo.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7970 SIP - few questions
Hi Omar, Where to dial *+*+#+*+*+# ? If I done it on settings menu, it unlocks the phone, and than again locks it... One more question. I have dialplan.xml from 7940 and 7960, can I use it with 7970? I have tried to define it like this dialTemplatedialplan.xml/dialTemplate But that doesn't work (dialplan.xml is in root of tftp). -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr Restarting the 7970 is like unlocking it twice, *-*-# to unlock, *-*-# to reboot. I don't believe hint functionality works on the SIP firmware for the 7970. Omar A. Sabek ___ --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
SV: [Asterisk-Users] treating an incoming call as a local extension
http://www.voip-info.org/wiki-Asterisk+cmd+DISA I think is what you are looking for :) -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av jnuoiqweahf kajhdsff Sendt: 27. april 2006 06:21 Til: asterisk-users@lists.digium.com Emne: [Asterisk-Users] treating an incoming call as a local extension I have [EMAIL PROTECTED] running on one machine, with X-lite running on another machine on my local network, with X-lite logged in to asterisk as extension 200. From X-lite, I can dial *97 to hear voice mail for extension 200, dial 201 to call extension 201, etc. I need to be able to accept an incoming call over the voip trunk which I have set up, and have asterisk treat that call as extension 202, so that e.g. I can dial in to asterisk from an external voip line and then as soon as asterisk answers the line, I can enter enter a password and then have the call treated as extension 202 and enter *97 to access the voicemail for 202, or enter 200 to locally call extension 200, etc. How can I do this? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Pattern matching problem
Yes, you are correct.I am so sorry. I never use the zap analog card. We only have one digium T1/E1 PCI card in our small office. One more question, The analogue zap channel is fxo port? Or fxs port? Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, April 27, 2006 9:10 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: Pattern matching problem On Wednesday 26 April 2006 20:54, kevin ling wrote: Same dial pattern on my extension.conf, But it's work great. The Asterisk only match 7 digits number. My * version is 2.1.6. From an analogue Zap channel? Bullshit. Analogue channels do not present the extension in one shot -- they present the digits one at a time, in sequence. When the dialplan matches, it matches. Why do you think the telco needs you to enter 1 for long distance? And why do you think they're moving to ten-digit dialing for so many areas? This is very very basic, standard pattern matching. Analogue channels are very different from digital ones in how the desired extension or telephone number is presented to the switch. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SATA hard disk compatibility
Hi! I have been using ASterisk 1.0.3 on Red hat Linux 9.0 for a long time now on my Home PC. I want to shift to a PC having SATA hard disk .Can I install Redhat 9.0 on SATA hard disk ??some people are telling me that I have to go for Linux Enterprise 4.0.I don`t want to leave Linux 9.0 because I want to run Asterisk 1.0.3 Can anyone help me?? Amna ___ --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] SMP kernel on Pent 4?
Rich Adamson wrote: Mike Fedyk wrote: Rich Adamson wrote: Had a Pent 4 server running fc3 crash (kernel panic) and am I then noticed that FreePBX installed using a SMP kernel (and grub indicated a non-SMP kernel was installed as well). Would running an SMP kernel on a Pent 4 potentially cause a kernel panic? (Or, do I need to dig somewhere else?) I remember that there were problems on FC running on P4 with HT (hyperthreading). If you have only one physical CPU, run something like 'top' or 'cat /proc/cpuinfo' which shows you how many CPU system use. If you have 1 fyz.CPU but 2 CPU in sys means P4-HT and SMP could be fine. You can try to switch off HT support in BIOS and run nonSMP kernel, and you'l see if it is better/more stable. I'm running 2pcs of dual core xeons, so I have 4 (logical) CPU. And I'm running FC4 with 2.6.16.-1.2096 SMP kernel. by xsilver ___ --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] Camp on?
I believe what you refer to is called Ring Back When Free at least thats how I know it in the UK. Ah yes, no I remember. We called it Automatic Ring Back. So we had normal ARB, or ARB on next use. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Pattern matching problem
Hi Andrew,
Sorry for my english first.
My configuration and hardware: AAH2.7 2.8, Digium TE100P, welltech 4fxo
voice gateway
SIP Phone
|
|
Asterisk Server - TE100P - Telcom1
|
+ Welltech 4FXO voicegateway Telcom2
Actually no matter on the digital interface (TE110P) or analog channels
(4FXO). Bellowing is my outbound routing config. I try to dial 6137451576
number. The asterisk doesn't match this dial pattern. And when I dail
6137451. It's work.
So you mean the analogue channels is analog phone attach on a fxs port?
[outrt-001-outside]
include = outrt-001-outside-custom
exten = _NXX,1,Macro(dialout-trunk,1,${EXTEN},,)
exten = _NXX,n,Macro(outisbusy,)
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, April 27, 2006 9:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re: Pattern matching problem
On Wednesday 26 April 2006 20:54, kevin ling wrote:
Same dial pattern on my extension.conf, But it's work great. The
Asterisk only match 7 digits number. My * version is 2.1.6.
From an analogue Zap channel? Bullshit.
Analogue channels do not present the extension in one shot -- they present
the digits one at a time, in sequence. When the dialplan matches, it
matches. Why do you think the telco needs you to enter 1 for long
distance? And why do you think they're moving to ten-digit dialing for so
many areas?
This is very very basic, standard pattern matching. Analogue channels are
very different from digital ones in how the desired extension or telephone
number is presented to the switch.
-A.
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On Wednesday 26 April 2006 20:34, hugolivude wrote:
Thanks, but the problem's with the first extension:
exten = _NXX,1,NoOp(Number dialed ${EXTEN}) exten =
_NXX,n,Dial(Zap/1/${EXTEN})
The problem is I _do_ get a match as you can see by the CLI output,
but it shouldn't match IMO - 6137451576 shouldn't match _NXX but
that line gets executed.
When you dial six one three seven four five one Asterisk says hey! That
matches _NXX! -- the fact that you have five seven six left means
nothing, just as you can dial 1-800-PROGRESSIVE as Eric stated earlier.
On analog Zap interfaces, Asterisk (just like the telco) simply listens
until the digits match. If you don't want a ten digit number to match, then
adjust your dialplan accordingly. This is not a strange error in Asterisk,
it is a mismatch between what you want the system to do and how the system
operates.
Digital Zap channels and VOIP channels do not work this way because the
entire number is sent in one go -- when you dial from a SIP phone,
Asterisk does not see a stream of digits, it sees one message or packet
of information with the entire phone number in it. That is why it doesn't
match with SIP or IAX or PRI channels. (overlap dial excepted.)
-A.
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Re: [Asterisk-Users] astcc: need partial pin code
Ronald Wiplinger wrote:
I have not used astcc with pin codes so far, since I set-up the phone
number as card number.
Some of my users want now to dial in to the system and than use their
card, which is their phone number.
For that I would need a way of authentication, like a pin.
I want to use something like:
What is your card number: user keys in the number
Enter your pin:user enter a long pin
Enter your destination phone number: user enters the destination
phone number
Is there a code snip available for that?
Keyin needs always more time, we need to allow longer spaces between
the digits, therefore we need to allow the # to finish the dialstring
faster. I wonder if we can use one dialstring for all:
cardnumber*pin*destination-number
How can a user end the call and dial a new number, without hanging up?
The user has usually a desk phone (=card number), and this dialin
should work parallel, but of course it assumes still that only one
card is in use.
bye
Ronald Wiplinger
I tried now the examples in the wiki, but they do not fit!!!
If I use in configure Require Pins Yes then everyone needs a pin code!
If I use in configure Require Pins NO then calling in people will just
need to know a valid card number!!!
How can I overcome this?
How can I re-write:
exten = _77.,1,Answer
exten = _77.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:2},3)
exten = _77.,3,Hangup
sothat the dialstring:
77*123456789012*159753*011886939775516 would be splitted into:
${CARDNUM}=123456789012
${PIN}=159753
${DESTINATION}=0118869397755516
with a mysql lookup of the cardnum in astcc get the pin and compare to
the given pin. If all is ok, than use the dial command
bye
Ronald Wiplinger
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[Asterisk-Users] Asterisk Hangs the whole system
Is it possible for asterisk to hang the whole system ?? My Linux box is acting up, and I want to be sure which way to look. Asterisk or some hardware. -- Regards, Nasir. ___ --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] AGI and incoming call
anyway, you could put the routing stuff in an external file included in extensions.conf: ... some dialplan stuff ... [extension-routing] #include ext-routing.conf ... some dialplan stuff ... in ext-routing.conf you have your routing stuff: exten = 1234567,1,Dial(11) exten = 7654321,1,Dial(12) and so on When you have a new customer, you only need to regenerate ext-routing.conf from the db and asterisk -rx extensions reload without having asterisk wait an AGI script to re-parse the routing stuff on each and every call... hope this helps 2006/4/26, Olivier Saulnier [EMAIL PROTECTED]: Hello,It's not possible, because the flat file is generated since a database,and each day, there is news customers. Best regards,Olivier S.Innocent Evil a écrit :Why don't you do something like this:exten = 12345678,1,Dial(10)exten = 45874521,1,Dial(11)exten = 32544884,1,Dial(12) replace Dial(10) and so on with apppriate extension.Thanks,--You don't have any choice, you already made it before you came here. -Original Message-From: [EMAIL PROTECTED]Sent: Wed, 26 Apr 2006 08:47:03 +0200To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] AGI and incoming callHello,I would like to intercept each incoming call and with an awk script,search the internal phone number ask. For example:I have a text database as this:External phone Internal Phone12345678 1045874521 1132544884 12 When the client 45874521 call, Asterisk must routed the incoming call tothe internal phone 11I have an awk script able to find the good internal phone, but i don'tknow how to interface it with Asterisk. I thought that AGI is the best way. Is it?Best regards,--Olivier SaulnierSTEGANUX35 Quai Louis Blanc03100 MontluçonT: 04.70.02.80.55 F: 04.70.02.80.57http://www.steganux.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Olivier SaulnierSTEGANUX35 Quai Louis Blanc03100 MontluçonT: 04.70.02.80.55F: 04.70.02.80.57http://www.steganux.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SATA hard disk compatibility
The Hardware support of SATA in RH9.0 is not fully integrated AFAIK , so moving to a SATA hard disk without an upgrade might not be the safest bet. on the other hand until you try you won't know for sure . have you thought of using the Fedora Core ? those have SATA support and they should be the closest thing to RH9 you can find. why don't you want to upgrade the asterisk ? 1.0.3 is a very old version and many fixes and features where added to the software . Assaf amna saleem wrote: Hi! I have been using ASterisk 1.0.3 on Red hat Linux 9.0 for a long time now on my Home PC. I want to shift to a PC having SATA hard disk .Can I install Redhat 9.0 on SATA hard disk ??some people are telling me that I have to go for Linux Enterprise 4.0.I don`t want to leave Linux 9.0 because I want to run Asterisk 1.0.3 Can anyone help me?? Amna ___ --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 -- Assaf Flatto Atelis IT Manager Cellular: +972-54-5679230 e-mail: [EMAIL PROTECTED] ___ --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] GXP-2000: disable provisioning
Hi, is there a way to completely disable TFTP/HTTP provisioning on the Grandstream GXP-2000? Thanks -- Domenico Viggiani ___ --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] getting asterisk to reliably answer a voip line
maybe you can try to issue a sip show registry on the console on a regular basis and watch if your * loose registration. You can also turn on sip debug on the console, to see if the unanswered calls effectively reach asterisk or not. In the latter, is sipphone that loose your registration, so you maybe can lower the time before registration renewals. And turn on qualify=yes for your peer to keep fresh nat mappings on the router. Search voip-info.org for more infos Hope this helps 2006/4/27, jnuoiqweahf kajhdsff [EMAIL PROTECTED]: I have a sipphone.com account, with asterisk set toanswer incoming calls, using the following settings (phone number and password omitted) in the PeerDetails for the SIP Trunk:allow=ulawcontext=from-pstndtmfmode=rfc2833fromdomain=proxy01.sipphone.com fromuser=1747xxxhost=proxy01.sipphone.cominsecure=verysecret=xtype=peerusername=1747xxxThe Asterisk machine is behind a Linksys router (full cone NAT).About 25% of the time, when I call that number (fromanother sipphone account), asterisk answers the line,but about 75% of the time, asterisk fails to answer,and doesn't even indicate that any incoming call was attempted, and sipphone times out after 15-20 secondsand dumps the unanswered call to its voicemail system.I don't see any pattern to the intermittent answering,and sometimes I can try numerous times and get no answer, and sometimes I can try several times in a rowand get an answer each time. It seems random. Outgoingcalls work 100%; only incoming are having problems.How can I diagnose whether the problem is with Asterisk or with Sipphone, or whether one or both arehaving problems because of NAT? Bypassing the NATrouter is not an option, even for testing. Is this aknown problem with Sipphone? How do the various voip providers (Sipphone, FWD, Broadvoice, etc) comparewith regards to incoming call completion reliabilitywhen the receiving device (Asterisk in this case) isbehind NAT?I'll eventually need to accept incoming PSTN calls via voip and I'm willing to pay for reliable service fromany provider, but I do need Asterisk to actuallyreceive and answer all attempted incoming calls.__ Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2.4/7 and chan_modem
Thanx, but for the record and archive purposes this did not work in 1.2.7.1 but it does work with 1.2.4. Marnus van Niekerk tom wrote: Marnus van Niekerk wrote: Hi, I am currently running several * boxes on 1.0.9 with HFC chipset ISDN modems using i4l's hisax driver and chan_modem. Will I be able to use my existing chan_modem setup with 1.2.4 or 1.2.7 or will I need to change it to use bristuff or chan_capi? I want to do the upgrade with as little changes as possible. Thank you Marnus van Niekerk -- chan_modem is still shipped with 1.2.7, you just need to uncomment line 21 in the Makefile that is in the channels folder of the source before compiling asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --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] Asterisk Voice Problems
Hi, I am running Asterisk 1.2.1 using Digium TDM 400P with 4FXO lines to connect to the PSTN world. But, I constantly get clipped voice whenever there is a call placed using Zap channels. I have tried it all the recommended solutions - turned off all non essential services on the machine - ran fxotune - Changed IRQ settings But nothing works. The only thing that works is reducing the rxgain to around -20. But this leads to other issues like the hangup on the PSTN line is not detected by Asterisk. Anyone have a clue about how to fix the bad quality problem. Any help will be highly appreciated. Thanks, Shyam ___ --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 Voice Problems
Hi, Have you try to install this TDM400P card on another asterisk server? Same problems? Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shyam Gopale Sent: Thursday, April 27, 2006 5:07 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Voice Problems Hi, I am running Asterisk 1.2.1 using Digium TDM 400P with 4FXO lines to connect to the PSTN world. But, I constantly get clipped voice whenever there is a call placed using Zap channels. I have tried it all the recommended solutions - turned off all non essential services on the machine - ran fxotune - Changed IRQ settings But nothing works. The only thing that works is reducing the rxgain to around -20. But this leads to other issues like the hangup on the PSTN line is not detected by Asterisk. Anyone have a clue about how to fix the bad quality problem. Any help will be highly appreciated. Thanks, Shyam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to Dial a number , after getting a mail notification ,
Hi I am looking for some advice or tips on how to make asterisk , to dial a number , when the asterisk server gets some mail to the asterisk user , Is it possible to do so Guidance requested Thanks Joseph John ___ Switch an email account to Yahoo! Mail, you could win FIFA World Cup tickets. http://uk.mail.yahoo.com ___ --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] AGI and incoming call
Hello, I thought it's exactly what i ask!! Very well!! Bets regards, Olivier S. picciuX a écrit : anyway, you could put the routing stuff in an external file included in extensions.conf: ... some dialplan stuff ... [extension-routing] #include ext-routing.conf ... some dialplan stuff ... in ext-routing.conf you have your routing stuff: exten = 1234567,1,Dial(11) exten = 7654321,1,Dial(12) and so on When you have a new customer, you only need to regenerate ext-routing.conf from the db and asterisk -rx extensions reload without having asterisk wait an AGI script to re-parse the routing stuff on each and every call... hope this helps 2006/4/26, Olivier Saulnier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello, It's not possible, because the flat file is generated since a database, and each day, there is news customers. Best regards, Olivier S. Innocent Evil a écrit : Why don't you do something like this: exten = 12345678,1,Dial(10) exten = 45874521,1,Dial(11) exten = 32544884,1,Dial(12) replace Dial(10) and so on with apppriate extension. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Sent: Wed, 26 Apr 2006 08:47:03 +0200 To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [Asterisk-Users] AGI and incoming call Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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 to Dial a number , after getting a mail notification ,
the most part will be to configure your MTA to trigger a script when the mail gets in. It depends on which MTA you're using. Once this is ok, you only have, from that script, to generate an auto-dial file to drop in asterisk spool directory to make it dial. 2006/4/27, John Joseph [EMAIL PROTECTED]: Hi I am looking for some advice or tips on how tomake asterisk , to dial a number , when the asterisk server gets some mail to the asterisk user , Is it possible to do soGuidance requested Thanks Joseph John___ Switch an email account to Yahoo! Mail, you could win FIFA World Cup tickets. http://uk.mail.yahoo.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Camp on?
On Wed, 2006-04-26 at 20:15 -0500, Eric ManxPower Wieling wrote: Something along the lines of show application retrydial ? Afaict RetryDial does not allow the caller to hang up the phone and wait for a call the moment the remote party hangs up. Any way to do this *without* the caller having to stay on the phone? Regards, Patrick ___ --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] Camp on?
On Thu, 2006-04-27 at 11:10 +0800, Nathan Alberti wrote: On 27/04/2006, at 9:15 AM, Eric ManxPower Wieling wrote: Something along the lines of show application retrydial ? [EMAIL PROTECTED] wrote: I am looking for that feature to implement on Asterisk as well. does anyone know how to implement it/ Thanks! - Original Message - From: Jon Farmer [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 26, 2006 12:09 PM Subject: Re: [Asterisk-Users] Camp on? I believe what you refer to is called Ring Back When Free at least thats how I know it in the UK. Not quite retrydial, that is more like dial the person and keep dialing the person until they pickup with the ability to listen to MOH while you wait. I think what the OP is asking for is call someone, they are busy, press a button to enable callback and as soon as they hang-up from a call the callers handset starts ringing to let them know the called party is now free, when the caller pickups the phone they are connected to the person they were trying to call. I'm not sure how clear my explanation is :S That explanation is fine :) And you are stressing the right point: the caller should be able to put down the phone after pressing 5. RetryDial does not seem to allow that. Perhaps it could be done in h with a DeadAGI but I have no idea how to do that. Suggestion examples most welcome. Regards, Patrick ___ --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] some EICON Diva 4BRI questions
Armin Schindler wrote: On Wed, 26 Apr 2006, Klaus Darilion wrote: On Sun, April 23, 2006 16:30, Armin Schindler said: On Sat, 22 Apr 2006, Klaus Darilion wrote: But I'm still confused. Usually, if a line needs termination, the termination is needed on both ends. Thus, if there is no line termination inside the DIVA card, we would termination in TE mode too. Sorry, I don't know why this is different to 'normal' ISDN line. Maybe the difference is about the distance. A normal ISDN line may be more than a few meters, but in NT-mode I have my PBX/Equipment connected to the card with a one meter cross-cable. I will try to find out more about this. Hi Armin! Could you find out something? Yes, I asked Eicon. The cards don't have internal termination. The ports are layed out for TE mode, so depending on your ISDN bus topology, you need a termination. And you are right, when in NT-mode both sides needs termination just normal ISDN bus topology. This still confuses me. In P2P mode, there is always one device in NT mode, and one device in TE mode. Usually termination is needed on both sides of the cable. If the EICON DIVA does not have termination resistors inside (IMO a real weakness, all the cheaper cards do have them and you can enable them with jumpers), I would think that we need external resistors in both cases, regardless if EICON DIVA is in NT or TE mode. Since I have only one termination in the middle of my cross cable, which works fine, I assume that the termination is not really necessary on both sides when having just a one meter cable. btw: do you know where I can by this cables (I do not like soldering)? Does EICON offer this cables (for fixing their design bug)? regards klaus ___ --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] Camp on?
On 27/04/2006, at 5:45 PM, Patrick wrote:
On Thu, 2006-04-27 at 11:10 +0800, Nathan Alberti wrote:
On 27/04/2006, at 9:15 AM, Eric ManxPower Wieling wrote:
Something along the lines of show application retrydial ?
[EMAIL PROTECTED] wrote:
I am looking for that feature to implement on Asterisk as well.
does anyone know how to implement it/
Thanks!
- Original Message - From: Jon Farmer
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 26, 2006 12:09 PM
Subject: Re: [Asterisk-Users] Camp on?
I believe what you refer to is called Ring Back When Free at
least thats how I know it in the UK.
Not quite retrydial, that is more like dial the person and keep
dialing the person until they pickup with the ability to listen to
MOH while you wait.
I think what the OP is asking for is call someone, they are busy,
press a button to enable callback and as soon as they hang-up from a
call the callers handset starts ringing to let them know the called
party is now free, when the caller pickups the phone they are
connected to the person they were trying to call.
I'm not sure how clear my explanation is :S
That explanation is fine :) And you are stressing the right point: the
caller should be able to put down the phone after pressing 5.
RetryDial does not seem to allow that. Perhaps it could be done in h
with a DeadAGI but I have no idea how to do that. Suggestion
examples
most welcome.
Regards,
Patrick
More appropriate may be a jump on BUSY, else the action may be
processed when not required i.e. after a successful call.
[macro-stdext]
exten = s,1,NoOp
exten = s,n,Dial(SIP/${ARG2},20)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1, --- DO STUFF HERE :)
; - Playback The person is busy, press 1 to have them call you back
when available or 2 to leave them a voicemail
; - Add code to catch 1 or 2 as appropriate and send to voicemail or
script to check channel status and handle call back.
exten = s-NOANSWER,1,Macro(voicemail,U${ARG3})
exten = s-NOANSWER,n,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
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RE: [Asterisk-Users] treating an incoming call as a local extension
kevin ling wrote: Check the DISA command. Yup, that does exactly what I need. Thanks! __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] astcc: need partial pin code
On Thursday 27 April 2006 11:08, Ronald Wiplinger wrote:
Ronald Wiplinger wrote:
I have not used astcc with pin codes so far, since I set-up the phone
number as card number.
Some of my users want now to dial in to the system and than use their
card, which is their phone number.
For that I would need a way of authentication, like a pin.
I want to use something like:
What is your card number: user keys in the number
Enter your pin:user enter a long pin
Enter your destination phone number: user enters the destination
phone number
Is there a code snip available for that?
Keyin needs always more time, we need to allow longer spaces between
the digits, therefore we need to allow the # to finish the dialstring
faster. I wonder if we can use one dialstring for all:
cardnumber*pin*destination-number
How can a user end the call and dial a new number, without hanging up?
The user has usually a desk phone (=card number), and this dialin
should work parallel, but of course it assumes still that only one
card is in use.
bye
Ronald Wiplinger
I tried now the examples in the wiki, but they do not fit!!!
If I use in configure Require Pins Yes then everyone needs a pin code!
If I use in configure Require Pins NO then calling in people will just
need to know a valid card number!!!
How can I overcome this?
How can I re-write:
exten = _77.,1,Answer
exten = _77.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:2},3)
exten = _77.,3,Hangup
sothat the dialstring:
77*123456789012*159753*011886939775516 would be splitted into:
${CARDNUM}=123456789012
${PIN}=159753
${DESTINATION}=0118869397755516
with a mysql lookup of the cardnum in astcc get the pin and compare to
the given pin. If all is ok, than use the dial command
Hi Ronald,
Just to give you an idea
I would suggest you to make two .agi files:
astcc.agi and astcc-disa.agi
In astcc.agi you'd leave everithing as it is, and enable
PIN =YES through the astcc-admin.cgi.
Thus you could dial without interogation:
exten = _1NXXNXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
astcc-disa.agi is a copy of astcc.agi so
# cp astcc.agi astcc-disa.agi.
# pico astcc-disa.agi.
Find the line:
# At this point we have a valid card number.
and coment out everything until:
# At this point we have a valid card and pin number.
You can dial from outside:
exten = 1234567894,1,DeadAGI(astcc-disa.agi)
and will de asked for cardnumber and pin.
Some mobile phones support w inside of a dialstring i.e.
1234567894w123456789012#w159753# .Fist part is the
DID you dial to enter * .
* asks for a cardnumber and the mobile waits for you on w
to pushEnter,
* asks for a pin and phone waits for you on w to push Enter
for the last string.
After all that you would here:Please enter the number you wish to dial...
Hope, this helps.
Benchev
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Re: [Asterisk-Users] some EICON Diva 4BRI questions
On Thu, 27 Apr 2006, Klaus Darilion wrote: Armin Schindler wrote: On Wed, 26 Apr 2006, Klaus Darilion wrote: On Sun, April 23, 2006 16:30, Armin Schindler said: On Sat, 22 Apr 2006, Klaus Darilion wrote: But I'm still confused. Usually, if a line needs termination, the termination is needed on both ends. Thus, if there is no line termination inside the DIVA card, we would termination in TE mode too. Sorry, I don't know why this is different to 'normal' ISDN line. Maybe the difference is about the distance. A normal ISDN line may be more than a few meters, but in NT-mode I have my PBX/Equipment connected to the card with a one meter cross-cable. I will try to find out more about this. Hi Armin! Could you find out something? Yes, I asked Eicon. The cards don't have internal termination. The ports are layed out for TE mode, so depending on your ISDN bus topology, you need a termination. And you are right, when in NT-mode both sides needs termination just normal ISDN bus topology. This still confuses me. In P2P mode, there is always one device in NT mode, and one device in TE mode. Usually termination is needed on both sides of the cable. Yes, but when you have a normal ISDN line (NT comes from Telco), then you have the termination in the NTBA (configurable with switches). If the EICON DIVA does not have termination resistors inside (IMO a real weakness, all the cheaper cards do have them and you can enable them with jumpers), I would think that we need external resistors in both cases, regardless if EICON DIVA is in NT or TE mode. That is not a weakness! The termination depends on your bus topology, so you should add the termination in the box at the wall when needed. Maybe some cards do have own termination on board, but this then need to be optional. I don't know any ISDN phone, which has own termination. Since I have only one termination in the middle of my cross cable, which works fine, I assume that the termination is not really necessary on both sides when having just a one meter cable. btw: do you know where I can by this cables (I do not like soldering)? Does EICON offer this cables (for fixing their design bug)? I'm not aware of such a cable to buy. Normaly, when you create a NT-side the connection is not made with just one cable (like I did because both device are just 10cm away from each other). In most cases you have an ISDN bus cabled in the rooms where the necessary changes (other termination, crossed) can be done in the boxes. I don't see any design bug here. Armin ___ --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 as a phone survey system
Unless things have changed, TeleYapper could only accomplish low volume one at a time calls. Kerry Garrison wrote: Asterisk is simply a telephony toolkit, so the simple answer is yes, Asterisk can do this. Also, being a toolkit means there are a number of ways to accomplish it. You could right PERL, Python, TCL, C, PHP or numerous other types of scripts that can manage this for you. To see how to do some of the basic functions, you can look at some of the scripts at Nerd Vittles (http://nerdvittles.com). Things like the TeleYapper will give you a basis to work from. Kerry Garrison Publisher - http://GeekGazette.com http://geekgazette.com/ - http://VOIPSpeak.net http://voipspeak.net/ (949) 502-7819 x200 - //[EMAIL PROTECTED]// mailto:[EMAIL PROTECTED] //http://www.techdatapros.com// http://www.techdatapros.com/ *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *TV JOE *Sent:* Wednesday, April 26, 2006 7:31 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] Asterisk as a phone survey system Hi, I'm interested in developing an automated phone survey and am curious if Asterisk could be configured to run such a system.. My idea is to record a message and a series of sub-questions. The system would call each number on a list and play the message, Depending on the touch tone response another message would be played. Is it possible for asterisk to manage a survey like this? If so can the responses from the listeners be recorded. If someone else has done this I'd be interested in details. TIA , TV JOE Yahoo! Messenger with Voice. http://us.rd.yahoo.com/mail_us/taglines/postman3/*http://us.rd.yahoo.com/evt=39666/*http://messenger.yahoo.com PC-to-Phone calls for ridiculously low rates. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Autodial feature doesn't return $DIALSTATUS values
Hello, I'm writing a small PHP application that generates calls automatically and tries to store call details on a Mysql Db, using manager API . When making an autodial call, I noticed that I couldn't read $DIALSTATUS values; since I can't evaluate dial status (BUSY, CONGESTION, NOANSWER), I can't understand when a receiver was busy or not. Nobody seems to have solved this problem; I visited many wiki sites but no answer was found. Could you help me to understand how could I get these values ? Thank you. _ INFINITO ADSLFLAT 4 MEGA: SOLO 27,90 EURO AL MESE IVA INCLUSA IP STATICO, BANDA GARANTITA 256Kbps ANTIVIRUS E FIREWALL INCLUSI NEL PREZZO http://adsl.infinito.it ___ --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] Help on chan_misdn and MSN's
Finally I'm not sure I found a small compatibility problem between chan_misdn and the Romanian implementation of ISDN or I simply solved a configuration problem with a huge hammer but I'm happy it works! You should try the following combination: immediate=no always_immediate=no --- Amatisoft SRL http://amatisoft.homelinux.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
--- rommel malana [EMAIL PROTECTED] wrote: Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel- Rommel, You should read the book Asterisk: The future of telephony (I believe is the name). There is a PDF of it available online. Do a google search and you should find it. Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] getting asterisk to reliably answer a voip line
--- picciuX wrote: maybe you can try to issue a sip show registry on the console on a regular basis and watch if your * loose registration. Ok: asterisk1*CLI sip show registry HostUsername Refresh State proxy01.sipphone.com:5060 17476510045105 Registered Also, at my.sipphone.com, when I log in and view advanced features, in SIP Registrations the status is always on line and Public IP address shows the IP address of the NAT device which my asterisk machine is behind, followed by e.g. (expires in 1020 seconds). According to both asterisk and sipphone, I'm never losing registration. You can also turn on sip debug on the console, to see if the unanswered calls effectively reach asterisk or not. I did sip debug on the console and got SIP Debugging enabled. Now, every 20 seconds or so, I get: -- SIP read from 192.168.3.22:5060: --- (0 headers 0 lines) Nat keepalive --- Trying to call after enabling debugging, some calls succeed and some fail (as usual), and I get no indication of the call attempts on the console when the call fails. Every minute or so, I get long spiels on the console (unrelated to the timing of my call attempts) starting with: REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 198.65.166.131:5060: REGISTER sip:proxy01.sipphone.com SIP/2.0 which sometimes contain strange things like: Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 192.168.3.22:5060: --- (0 headers 0 lines) Nat keepalive --- asterisk1*CLI -- SIP read from 198.65.166.131:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.23:5060;branch=z5lR3bK653e4bb;rport=1937;received =72.171.58.49 From: sip:[EMAIL PROTECTED];tag=as741dda96 To: sip:[EMAIL PROTECTED];tag=21a68532c2cd5d9b34affe6bba40a2e. 1bb5 Call-ID: [EMAIL PROTECTED] CSeq: 166 REGISTER P-Behind-NAT: Yes Contact: sip:[EMAIL PROTECTED]:1936;q=0.00;expires=26 Contact: sip:[EMAIL PROTECTED]:1937;q=0.00;expires=120 Content-Length: 0 --- (10 headers 0 lines)--- Scheduling destruction of call '[EMAIL PROTECTED] ne.com' in 32000 ms Which is strange because I had no incoming or outgoing calls or call attempts at the time I got those messages on the console, yet asterisk is talking about destroying calls. In the latter, is sipphone that loose your registration, Yes, this appears to be the case. so you maybe can lower the time before registration renewals. But during the time I was doing tests and recording the above information to put in this message, I had several call attempts succeed and several fail, and several minutes later, the SIP registration I mentioned at my.sipphone.com was down from 1020 seconds to 681 seconds, and then later I checked again and it was down to 412 seconds, etc. So all the while when I was having some calls succeed and some fail, my sipphone registration had not yet been renewed (according to sipphone). So I don't know what all the registration stuff is that asterisk is dumping to the console in debug mode, but it's apparently not reregistration with sipphone, since sipphone's timer doesn't get reset by it, and it doesn't seem to have any relationship to whether my incoming call attempts succeed or fail. And turn on qualify=yes for your peer to keep fresh nat mappings on the router. I tried that yesterday, and it seemed to have no effect. Based on this information, can you give any clue as to what the problem might be? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Anyone using the GSMgateway from CyberTelecom ?
I was going to buy two units from them. Seeing how everyone here talks about them I never went thru with it. Reputation caries a lot of weight. --- Benchev [EMAIL PROTECTED] wrote: Thanks Adibar, (sorry List:-) You have at least an offer. The only thing I've got so far was a promiss to have a decision tomorrow .For a month now, tomorrow is every day after the day before tomorrow. However, I hope, that in order to keep it's good reputation CyberTelecom will consider FREE OF CHARGE EXCHANGE of the useless GSM boxes or do immediate refund or something. Let's see... Cheers, Benchev On Wednesday 26 April 2006 01:17, adibar wrote: Hi Benchev News from the front. Sam is kinda offering me an exchange of my box. But I should return it to him at my cost ;-) Last word is not spoken yet on that, cause I'm really not amused on this ;-) Keeping you in loop. Greets Adibar On Mon, Mar 27, 2006 at 10:05:01AM +0300, Benchev wrote: Actually I've got five, but the first one I have received around Xmas and I don't have these problems with it. I use spa3000 as FXO and the gsm gateway works seamlessly inbound, outbound, DISA, no annoying sounds, no DTMF problems. There is one problem however, the gateway does not transfer correctly the CID to the FXO(at least in my case) but this could be a sipura problem as well. Now, the other 4 seam to be a different model or something and one should be very careful ordering that thing since you never know which model you are going to receive. They are used with no-brand-name FXO/FXS ATAs but I don't think that the ATAs is the problem. Everything goes wrong when the gateway is tested as a dock-n-talk (dialing through it connected to one of the RJ11 with an ordinary phone set). First there is no DTMF recognition whatsoever, and second tha gateway does not sense the hangup and start making the noises. Hope Sam could solve the problem with the factory or exchange the goods with working ones. Benchev Outch... Four of them and not working... That hurts. How do you connect them to * ? As I'm using only one for me an X100P-FXO is sufficiant and seems to work as good as attaching a real anlog phone. Btw. I saw that www.voipsolutions.be is selling them also, but for 165.- euro On Sun, Mar 26, 2006 at 03:28:07PM +0200, Benchev wrote: Hi Adibar, Thanks very much for the answer. We are also struggling (with 4 of them :( ) and will let you know how the things develop, too, in case of success. Have a nice Sunday, Benchev Hi Benchev I'm still in contact with Sam, but currently no changes. The device is still in an unusable state for me, as it only allows one call, which results in wild-beeping on terminating the call. But I still hope, that Sam finds anywhere a tech-person who can hand me out the correct setup-information. As soon as I get it in a working state, I will let you know it ;-) Adibar On Sat, Mar 25, 2006 at 09:55:56PM +0200, Benchev wrote: Hi Adibar, Any success with the gsm gateway? I have exactly the same problem with units received this month. The codes given by Sam are not working... Please, let me know if you have discovered something. Thanks in advance, Benchev But these are the wrong instructions again. Same as those ones you sent me allready. I've got the small box for £60 The only reaction I get is if I press just *. Then the display changes to SET___. After that there is silence for about 15 seconds. Pressing any keys is only allowd up to four digits. So also the given password is to long for entering. After the 15 seconds or the four digits I get a busy-signal. No password-prompt, no LINE CON. Nada... Adibar On Sat, Mar 11, 2006 at 05:12:40AM +0800, Sam Tam wrote: Hello To solved the beeping problem you need to first enter the configuration mode. I .Entry into SETTING STATUS 1) Pick up the phone, press the button of 0 ** #; 2) Screen display: ?SETUP?; Input pass: input pass word: 332808 Then will display IMPUT CON you can change the box working mode . use the command *000100#0#for set defaut ,billing mode. *000100#1#for one long tone mode *000100#2#for long tone mode Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of adibar Sent: Saturday,
Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?
I was going to buy two units from them. Seeing how everyone here talks about them I never went thru with it. Reputation caries a lot of weight. --- Benchev [EMAIL PROTECTED] wrote: Thanks Adibar, (sorry List:-) You have at least an offer. The only thing I've got so far was a promiss to have a decision tomorrow .For a month now, tomorrow is every day after the day before tomorrow. However, I hope, that in order to keep it's good reputation CyberTelecom will consider FREE OF CHARGE EXCHANGE of the useless GSM boxes or do immediate refund or something. Let's see... Cheers, Benchev On Wednesday 26 April 2006 01:17, adibar wrote: Hi Benchev News from the front. Sam is kinda offering me an exchange of my box. But I should return it to him at my cost ;-) Last word is not spoken yet on that, cause I'm really not amused on this ;-) Keeping you in loop. Greets Adibar On Mon, Mar 27, 2006 at 10:05:01AM +0300, Benchev wrote: Actually I've got five, but the first one I have received around Xmas and I don't have these problems with it. I use spa3000 as FXO and the gsm gateway works seamlessly inbound, outbound, DISA, no annoying sounds, no DTMF problems. There is one problem however, the gateway does not transfer correctly the CID to the FXO(at least in my case) but this could be a sipura problem as well. Now, the other 4 seam to be a different model or something and one should be very careful ordering that thing since you never know which model you are going to receive. They are used with no-brand-name FXO/FXS ATAs but I don't think that the ATAs is the problem. Everything goes wrong when the gateway is tested as a dock-n-talk (dialing through it connected to one of the RJ11 with an ordinary phone set). First there is no DTMF recognition whatsoever, and second tha gateway does not sense the hangup and start making the noises. Hope Sam could solve the problem with the factory or exchange the goods with working ones. Benchev Outch... Four of them and not working... That hurts. How do you connect them to * ? As I'm using only one for me an X100P-FXO is sufficiant and seems to work as good as attaching a real anlog phone. Btw. I saw that www.voipsolutions.be is selling them also, but for 165.- euro On Sun, Mar 26, 2006 at 03:28:07PM +0200, Benchev wrote: Hi Adibar, Thanks very much for the answer. We are also struggling (with 4 of them :( ) and will let you know how the things develop, too, in case of success. Have a nice Sunday, Benchev Hi Benchev I'm still in contact with Sam, but currently no changes. The device is still in an unusable state for me, as it only allows one call, which results in wild-beeping on terminating the call. But I still hope, that Sam finds anywhere a tech-person who can hand me out the correct setup-information. As soon as I get it in a working state, I will let you know it ;-) Adibar On Sat, Mar 25, 2006 at 09:55:56PM +0200, Benchev wrote: Hi Adibar, Any success with the gsm gateway? I have exactly the same problem with units received this month. The codes given by Sam are not working... Please, let me know if you have discovered something. Thanks in advance, Benchev But these are the wrong instructions again. Same as those ones you sent me allready. I've got the small box for £60 The only reaction I get is if I press just *. Then the display changes to SET___. After that there is silence for about 15 seconds. Pressing any keys is only allowd up to four digits. So also the given password is to long for entering. After the 15 seconds or the four digits I get a busy-signal. No password-prompt, no LINE CON. Nada... Adibar On Sat, Mar 11, 2006 at 05:12:40AM +0800, Sam Tam wrote: Hello To solved the beeping problem you need to first enter the configuration mode. I .Entry into SETTING STATUS 1) Pick up the phone, press the button of 0 ** #; 2) Screen display: ?SETUP?; Input pass: input pass word: 332808 Then will display IMPUT CON you can change the box working mode . use the command *000100#0#for set defaut ,billing mode. *000100#1#for one long tone mode *000100#2#for long tone mode Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of adibar Sent: Saturday,
Re: [Asterisk-Users] Asterisk as a phone survey system
Hi, I'm interested in developing an automated phone survey and am curious if Asterisk could be configured to run such a system.. My idea is to record a message and a series of sub-questions. The system would call each number on a list and play the message, Depending on the touch tone response another message would be played. Is it possible for asterisk to manage a survey like this? If so can the responses from the listeners be recorded. If someone else has done this I'd be interested in details. TIA , TV JOE Yes you can create this. It may be a headache but it can be done. To do it the old fashioned way you can create lots of diffrent context's and have the user be sent from one to another based on what they press. You can also have the results dumped in to a db. Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 Hangs the whole system
Is it possible for asterisk to hang the whole system ?? My Linux box is acting up, and I want to be sure which way to look. Asterisk or some hardware. People in the past had the problem. I dont remember what the cause of the problem was. Try looking at the archives. Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] zt_pri-error
hi all,I just installed Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1o and have a strange Warning in CLI, which is:WARNING[875]:chan_zap.c:8498 zt_pri_error: 1 TEI remove TEI = 0and another one: WARNING[875]: chan_zap.c:8498 zt_pri_error: 1 updating callstate, peercallstate 2 to 1Does anybody know what that could be? The first Warnings come every 10 minutes. I googled around and could not find anything. my zapata.conf:switchtype = euroisdn; p2p TE modesignalling = bri_cpepridialplan=localfaxdetect=bothrxgain=1.5txgain=1.5echocancel=yesimmediate=nooverlapdial=yesgroup = 1 context=isdnchannel = 1-2my zaptel.conf:loadzone=nldefaultzone=nlspan=1,1,3,ccs,amibchan=1-2dchan=3System: Slackware 10.0 Linux 2.4.32 with 1 HFC-S card (cologne). Thanks in advance Christian Gansberger ___ --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 IVR / Scalability
i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there any large scale deployment of asterisk to date ? Lots of people are using asterisk in a production enviroment. When you say IVR system what do you mean ? A GUI that will help you create it ? You can create your own IVR as long as you learn how to write a proper dial plan. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Accessing PARKEDAT variable in AGI
On Thursday 27 April 2006 00:25, jnuoiqweahf kajhdsff wrote: I'm attempting to do this in an AGI program: I have had *great* difficulty accessing channel variables in *ANY* AGI language for some time now. I have not filed a bug though, so I am partly to blame for its not being fixed. -A. ___ --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] Camp on?
On Thu, 2006-04-27 at 18:08 +0800, Nathan Alberti wrote:
On 27/04/2006, at 5:45 PM, Patrick wrote:
On Thu, 2006-04-27 at 11:10 +0800, Nathan Alberti wrote:
On 27/04/2006, at 9:15 AM, Eric ManxPower Wieling wrote:
Something along the lines of show application retrydial ?
[EMAIL PROTECTED] wrote:
I am looking for that feature to implement on Asterisk as well.
does anyone know how to implement it/
Thanks!
- Original Message - From: Jon Farmer
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 26, 2006 12:09 PM
Subject: Re: [Asterisk-Users] Camp on?
I believe what you refer to is called Ring Back When Free at
least thats how I know it in the UK.
Not quite retrydial, that is more like dial the person and keep
dialing the person until they pickup with the ability to listen to
MOH while you wait.
I think what the OP is asking for is call someone, they are busy,
press a button to enable callback and as soon as they hang-up from a
call the callers handset starts ringing to let them know the called
party is now free, when the caller pickups the phone they are
connected to the person they were trying to call.
I'm not sure how clear my explanation is :S
That explanation is fine :) And you are stressing the right point: the
caller should be able to put down the phone after pressing 5.
RetryDial does not seem to allow that. Perhaps it could be done in h
with a DeadAGI but I have no idea how to do that. Suggestion
examples
most welcome.
Regards,
Patrick
More appropriate may be a jump on BUSY, else the action may be
processed when not required i.e. after a successful call.
[macro-stdext]
exten = s,1,NoOp
exten = s,n,Dial(SIP/${ARG2},20)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1, --- DO STUFF HERE :)
; - Playback The person is busy, press 1 to have them call you back
when available or 2 to leave them a voicemail
; - Add code to catch 1 or 2 as appropriate and send to voicemail or
script to check channel status and handle call back.
exten = s-NOANSWER,1,Macro(voicemail,U${ARG3})
exten = s-NOANSWER,n,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
Thanks for the pointer Nathan. I slapped something together quick 'n
dirty but this is not working. The problem is that the call file is only
generated when the originating caller stays on the phone until the
remote caller hangs up. Suggestions how to fix this much appreciated.
extensions.conf (ARB = Automatic Ring Back)
exten = 35003,1,Set(ARBFROMEXT=${CALLERIDNUM})
exten = 35003,n,Set(ARBFROMCHAN=${CUT(CHANNEL|-|1)})
exten = 35003,n,Set(ARBTOEXT=${EXTEN})
exten = 35003,n,NoOP(ARBFROMEXT IS ${ARBFROMEXT})
exten = 35003,n,NoOP(ARBFROMCHAN IS ${ARBFROMCHAN})
exten = 35003,n,NoOP(ARBTOEXTIS ${ARBTOEXT})
exten = 35003,n,Dial(ZAP/3/,15,tT)
exten = 35003,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Congestion
exten = s-BUSY,1,Answer
exten = s-BUSY,n,Wait(1)
exten = s-BUSY,n,Background(extension)
exten = s-BUSY,n,SayNumber(${ARBTOEXT})
exten = s-BUSY,n,Background(is-curntly-busy)
exten = s-BUSY,n,Background(press-5)
exten = s-BUSY,n,Background(or)
exten = s-BUSY,n,Background(hangup-try-again)
exten = s-BUSY,n,WaitExten(5)
exten = 5,1,DBput(ARB/${CALLERIDNUM}=ON)
exten = 5,n,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
exten = h,1,DBGet(temp=ARB/${CALLERIDNUM})
exten = h,n,GotoIf($[${DBGETSTATUS} = FOUND]?found:bail)
exten = h,n(found),NoOP(h: ARB IS ON FOR ${CALLERIDNUM})
exten = h,n,DBdel(ARB/${CALLERIDNUM})
exten = h,n,Wait(5)
exten = h,n(check),ChanIsAvail(ZAP/3,s)
exten = h,n,NoOP(AVAILCHAN IS ${AVAILCHAN})
exten = h,n,NoOP(AVAILORIGCHAN IS ${AVAILORIGCHAN})
exten = h,n,NoOP(AVAILSTATUS IS ${AVAILSTATUS})
exten = h,n,GotoIf($[${AVAILSTATUS} = 2]?busy:callit)
exten = h,n(busy),Wait(5)
exten = h,n,Goto(check)
exten = h,n(callit),NoOP(h: ${ARBTOEXT} IS FREE. START AGI)
exten = h,n,DeadAGI(arb.agi,${ARBFROMEXT}|${ARBFROMCHAN}|${ARBTOEXT})
exten = h,n(bail),Hangup
*
arb.agi - using the automatic ring back name for this feature
*
#!/usr/bin/php -q
?php
ob_implicit_flush(true);
set_time_limit(6);
//$err=fopen(php://stderr,w);
$in = fopen(php://stdin,r);
$stdlog = fopen(/var/log/asterisk/my_agi.log, w);
function read() {
global $in, $debug;
$input = str_replace(\n, , fgets($in, 4096));
return $input;
}
function errlog($line) {
global $err;
echo VERBOSE \$line\\n;
}
function write($line) {
global $debug;
echo $line.\n;
}
// parse agi headers into array
while ($env=read()) {
$env = str_replace(\,,$env);
Re: [Asterisk-Users] billing realtime
JP Carballo wrote: Yes, certainly, through deadagi. I just have one question though, why reinvent the wheel? There are prepaid systems that work with asterisk. I have yet to find a prepaid system that allows multiple concurrent calls per account. Most seem to be based on a pin number also which I don't want. Anyone know of a system that allows concurrent calls? A while back some one posted some code that he used that took out the flag in astcc that kept track if there was a call in progress for that pin or not. Dont know if it wil work for real time though. Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Billing Server Open Source
astcc. it comes with asterisk. --- [EMAIL PROTECTED] wrote: Any know of any working smart open source billing? Something smart that can do prepay/postpay and disconnect customers when they owe or a disconnect a call in progress for low balance. ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Pattern matching problem
On Thursday 27 April 2006 03:30, kevin ling wrote: One more question, The analogue zap channel is fxo port? Or fxs port? Analog Zap channel or more generally, Analog channel (since chan_modem, chan_phone, and likely chan_mgcp too) means any channel technology which does NOT present the extension as a single message. Any technology which streams the numbers as they come when entering the dialplan will behave this way. Most digital technologies (SIP, IAX, PRI, etc.) present the extension as a single message, so Asterisk sees 10 digits or 5 digits or however many digits you dial all at once. Clear as mud? :-) -A. ___ --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] Camp on?
Andreas Sikkema wrote: I believe what you refer to is called Ring Back When Free at least thats how I know it in the UK. Ah yes, no I remember. We called it Automatic Ring Back. So we had normal ARB, or ARB on next use. Over the years, traditional pbx manufacturers have implemented multiple forms of camp-on and ring-back-when-free, and their marketing people tend to invent different terms for the same basic functionality. Implementing either one in asterisk needs to consider other extension config info such as whether the extension has VM or not, whether the call came from the pstn or another local extension, etc. Not likely either form can truly be implemented in a agi script without significantly impacting other pbx functions. ___ --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] astcc: need partial pin code
Hi Ronald,
Small mistake, see bellow:
Benchev
Just to give you an idea
I would suggest you to make two .agi files:
astcc.agi and astcc-disa.agi
In astcc.agi you'd leave everithing as it is, and enable
PIN =YES through the astcc-admin.cgi.
Thus you could dial without interogation:
exten = _1NXXNXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
astcc-disa.agi is a copy of astcc.agi so
# cp astcc.agi astcc-disa.agi.
# pico astcc-disa.agi.
pico astcc.agi.
Find the line:
# At this point we have a valid card number.
and coment out everything until:
# At this point we have a valid card and pin number.
You can dial from outside:
exten = 1234567894,1,DeadAGI(astcc-disa.agi)
and will de asked for cardnumber and pin.
Some mobile phones support w inside of a dialstring i.e.
1234567894w123456789012#w159753# .Fist part is the
DID you dial to enter * .
* asks for a cardnumber and the mobile waits for you on w
to pushEnter,
* asks for a pin and phone waits for you on w to push Enter
for the last string.
After all that you would here:Please enter the number you wish to dial...
___
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Re: [Asterisk-Users] SMP kernel on Pent 4?
Tomas Stribrny wrote: Rich Adamson wrote: Mike Fedyk wrote: Rich Adamson wrote: Had a Pent 4 server running fc3 crash (kernel panic) and am I then noticed that FreePBX installed using a SMP kernel (and grub indicated a non-SMP kernel was installed as well). Would running an SMP kernel on a Pent 4 potentially cause a kernel panic? (Or, do I need to dig somewhere else?) I remember that there were problems on FC running on P4 with HT (hyperthreading). If you have only one physical CPU, run something like 'top' or 'cat /proc/cpuinfo' which shows you how many CPU system use. If you have 1 fyz.CPU but 2 CPU in sys means P4-HT and SMP could be fine. You can try to switch off HT support in BIOS and run nonSMP kernel, and you'l see if it is better/more stable. I'm running 2pcs of dual core xeons, so I have 4 (logical) CPU. And I'm running FC4 with 2.6.16.-1.2096 SMP kernel. This is a single P4 (no dual core), and HT was turned on. Just turned it off now and still running non-smp kernel, so we'll see where stability goes from here. Kind of odd since the box had been running for over six months with fc3 and HT on. ___ --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] Interesting Dial-Plan Question
Hi, When I setup a user, I give them an extension like 570xxx. This is fine and dandy while in one area code, but we've since gone to other area codes.I'd like the user's to retain the ability to dial 7 digits no matter what number they have. Any thoughts on how to do that? EXAMPLE: User has number 7175551212. I want that when they dial 323 it dials 717-323-.User has number 5705551212. I want that when they dial 323 it dials 570-323-. I'm thinking I need to do some chopping, splitting, and variable writing?Am I correct in thinking that if I do all of this before my current dialplan kicks in that I should be able to still use my routing? IE: User dials 323 and is from 717 area code. If I transform the number to be 717323 the dial-plan will now read it as if the user punched it in that way and process it? ___ --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 Hangs the whole system
A.R. Nasir Qureshi wrote: Is it possible for asterisk to hang the whole system ?? My Linux box is acting up, and I want to be sure which way to look. Asterisk or some hardware. Both are possible. If you watched the cvs/svn commits over the last year or so, several asterisk issues have been identified and corrected relating to mem allocation, dereferencing, etc, etc. I don't know that anyone has actually kept track of bugs vs versions to know which versions might be suspect, but it might help if you'd include which distro/kernel you're running, asterisk version, types of cards installed, etc. You might also try running memtest just to rule out memory failures or issues. ___ --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] URGENTS: seek people for video tests with asterisk/ser/rtpproxy + eyebeam
Hi asterisk, openser, ser users. I have to check video support between asterisk, open(ser) and rtpproxy . ASTERISK (b2bua+registrar server) | | | | SER + rtpproxy | | NAT | | sip agents (with video support) Both signalling and media channels are kept in the path of SER+rtpproxy and ASTERISK . I can provide accounts to people who wish to test video with eyebeam for example ? Regards harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --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] Codec G729 / x86_64 bits.
how much are the codecs thst you cant buy em ? i dont intend to play judge and jurry however asterisk is a present that was given to all of us. im some way or another we should give to those that gave us. --- Jefferson Carvalho [EMAIL PROTECTED] wrote: Thanks for the suggestion , But I post a message to get a FREE codec (OPEN) , and not a PURCHASED. If I was interested in get a licensed one , believe that I never had Post a message on this list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Sunday, April 23, 2006 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Codec G729 / x86_64 bits. Jefferson Carvalho wrote: I always used a compiled version for a x86 system From [...] Someone could help me on this? Yes, the folks at Digium will be more than happy to help you. Visit http://www.digium.com/en/products/voice/g729codec.php and get a licensed codec. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to accept incoming PSTN calls
you have all these includes in your (messy) dial plan yet you didnt post the files that you use in include. --- Johnny Stork [EMAIL PROTECTED] wrote: I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running [EMAIL PROTECTED] 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on the Grandstream or any registered SIP sofware phone from any computer. I can also get a dial tone from the analog phone connected to the ZAP X100P port. But when incoming callas come in, none of the phones ring. No VoIP trunks, just the single ZAP trunk from the X100P. Below are my configurations and a tail of /var/log/asterisk/full when making a call from an outside line. There is much more in the extensions.conf file but I was not sure how much to include and noticed in another post that only a couple sections were included. Also, when making an outside PSTN call comes in the other non-asterisk-connected phones in the house ring fine, but none of the asterisk-connected extensions/phones? sip.conf file: [general] bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf #include additional_a2billing_sip.conf extensions.conf: zapata.conf file: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf extensions.conf file: ; include extension contexts generated from AMP #include extensions_additional.conf ; Customizations to this dialplan should be made in extensions_custom.conf ; See extensions_custom.conf.sample for an example #include extensions_custom.conf [from-trunk] ; just an alias since VoIP shouldn't be called PSTN include = from-pstn [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) var/log/asterisk/full (when recieving a call from pstn): Apr 26 18:43:33 VERBOSE[2696] logger.c: -- Remote UNIX connection Apr 26 18:43:52 VERBOSE[25804] logger.c: -- Remote UNIX connection disconnected Apr 26 18:44:57 VERBOSE[25810] logger.c: -- Starting simple switch on 'Zap/1-1' Apr 26 18:44:59 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:00 VERBOSE[25810] logger.c: -- Executing Goto(Zap/1-1, from-pstn|s|1) in new stack Apr 26 18:45:00 VERBOSE[25810] logger.c: -- Goto (from-pstn,s,1) Apr 26 18:45:00 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:00 DEBUG[2775] manager.c: Manager received command 'Command' Apr 26 18:45:00 DEBUG[2775] manager.c: Manager received command 'Command' Apr 26 18:45:01 VERBOSE[25810] logger.c: -- Executing Goto(Zap/1-1, from-pstn|s|1) in new stack Apr 26 18:45:01 VERBOSE[25810] logger.c: -- Goto (from-pstn,s,1) Apr 26 18:45:01 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:01 DEBUG[25810] chan_zap.c: Exception on 17, channel 1 Apr 26 18:45:01 DEBUG[25810] chan_zap.c: Got event Ring Begin(1Cool on channel 1 (index 0) Apr 26 18:45:02 VERBOSE[25810] logger.c: -- Executing Goto(Zap/1-1, from-pstn|s|1) in new stack Apr 26 18:45:02 VERBOSE[25810] logger.c: -- Goto (from-pstn,s,1) Apr 26 18:45:02 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:03 VERBOSE[25810] logger.c: -- Executing Goto(Zap/1-1, from-pstn|s|1) in new stack Apr 26 18:45:03 VERBOSE[25810] logger.c: -- Goto (from-pstn,s,1) Apr 26 18:45:03 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:03 DEBUG[25810] chan_zap.c: Exception on 17, channel 1 Apr 26 18:45:03 DEBUG[25810] chan_zap.c: Got event
Re: [Asterisk-Users] Interesting Dial-Plan Question
Matt wrote:
Hi,
When I setup a user, I give them an extension like 570xxx. This
is fine and dandy while in one area code, but we've since gone to
other area codes.I'd like the user's to retain the ability to dial
7 digits no matter what number they have. Any thoughts on how to do
that?
EXAMPLE: User has number 7175551212. I want that when they dial
323 it dials 717-323-.User has number 5705551212. I
want that when they dial 323 it dials 570-323-.
I'm thinking I need to do some chopping, splitting, and variable
writing?Am I correct in thinking that if I do all of this before
my current dialplan kicks in that I should be able to still use my
routing?
IE: User dials 323 and is from 717 area code. If I transform the
number to be 717323 the dial-plan will now read it as if the user
punched it in that way and process it?
exten = _NXX,1,Dial(Zap/g1/${CALLERIDNUM::0:3}${EXTEN})
This assumes that you set the user's Caller*ID number to be their
telephone number. It takes the first 3 digits of their CALLERIDNUM and
prepends it to the number they dialed.
See README.variables.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
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[Asterisk-Users] Transfer - context/priority
Hi list! When I'm doing transfer, to what context/priority does that call goes? Can it be changed? Is it the same for blind_tr/att_tr/and for transfer that appears when phone replies with - 302 Moved Temporarily? The thing is that I'm trying to transfer incoming call from E1 interface back to E1 interface. Transfers will occur when user is going out and sets up all call forward to his mobile. The problem is that I need to do something with the call (change caller ID) before I transfer it out. How can I achieve this? Thank you! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] TE405P vs. SoundCard problem
My Asterisk server believes that a Digium TE405P and sound are incompatible. Basically, no matter what else I do to the machine it terms of hardware, if the TE405P is installed, none of the playback/background/etc commands work. MOH works fine. So far, I have tried: 1) Seven different PCI soundcards with different chipsets. (Go go Computer Junk Store!) 2) Disabling each and every device offered by the motherboard, except the IDE and video. 3) Each possible PCI slot combination via trial and error. 4) Calling Digium, who while helpful, did not know how to solve the problem. '/proc/interrupts' usually likes to put the 'wct4xxp' on the same line as my 'Intel ICH5', but as of this moment that isn't the case. The sound card, by the way, exists for overhead paging purposes. My questions are: 1) Is there a document I should be aware of? 2) Has anyone else resolved this sort of problem before? 3) Do I need to scrap the sound card and use an FX(O) device? If so, how do I get my sound back? Thanks, Bob McDowell *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. ___ --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] Interesting Dial-Plan Question
Eric,
Yes.. I am setting calleridnum to be their phone number. And your
example is peachy... except for the fact that it assumes I want to go
out ZAP/g1!!
My problem is I have a very intricite routing plan that routes that
call out several different carriers depending on what you dialed.
(Long Distance, international, local, etc).
The way it works now is the dialplan just looks at the number you
dialed and routes based on that. I guess what I am asking is in
theory I should be able to do:
Look at origination number. Take first 3 digits and put into
variable. So 5705551212 becomes 570 in ${AREACODE}.
Now, look at the number we dialed. If it is (and this is where I am a
little unclear on what to do) 7 digits long then we append the
${AREACODE} variable. Else, we send it through to the dialplan as
is.
exten = _NXX,1,Dial(Zap/g1/${CALLERIDNUM::0:3}${EXTEN})
This assumes that you set the user's Caller*ID number to be their
telephone number. It takes the first 3 digits of their CALLERIDNUM and
prepends it to the number they dialed.
See README.variables.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
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Re: [Asterisk-Users] TE405P vs. SoundCard problem
Bob - what type of server/mobo are you using? Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Bob McDowell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 27, 2006 8:46 AM Subject: [Asterisk-Users] TE405P vs. SoundCard problem My Asterisk server believes that a Digium TE405P and sound are incompatible. Basically, no matter what else I do to the machine it terms of hardware, if the TE405P is installed, none of the playback/background/etc commands work. MOH works fine. So far, I have tried: 1) Seven different PCI soundcards with different chipsets. (Go go Computer Junk Store!) 2) Disabling each and every device offered by the motherboard, except the IDE and video. 3) Each possible PCI slot combination via trial and error. 4) Calling Digium, who while helpful, did not know how to solve the problem. '/proc/interrupts' usually likes to put the 'wct4xxp' on the same line as my 'Intel ICH5', but as of this moment that isn't the case. The sound card, by the way, exists for overhead paging purposes. My questions are: 1) Is there a document I should be aware of? 2) Has anyone else resolved this sort of problem before? 3) Do I need to scrap the sound card and use an FX(O) device? If so, how do I get my sound back? Thanks, Bob McDowell *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interesting Dial-Plan Question
On Thu, Apr 27, 2006 at 07:39:58AM -0500, Eric ManxPower Wieling scribbled:
Matt wrote:
Hi,
When I setup a user, I give them an extension like 570xxx. This
is fine and dandy while in one area code, but we've since gone to
other area codes.I'd like the user's to retain the ability to dial
7 digits no matter what number they have. Any thoughts on how to do
that?
EXAMPLE: User has number 7175551212. I want that when they dial
323 it dials 717-323-.User has number 5705551212. I
want that when they dial 323 it dials 570-323-.
I'm thinking I need to do some chopping, splitting, and variable
writing?Am I correct in thinking that if I do all of this before
my current dialplan kicks in that I should be able to still use my
routing?
IE: User dials 323 and is from 717 area code. If I transform the
number to be 717323 the dial-plan will now read it as if the user
punched it in that way and process it?
exten = _NXX,1,Dial(Zap/g1/${CALLERIDNUM::0:3}${EXTEN})
This assumes that you set the user's Caller*ID number to be their
telephone number. It takes the first 3 digits of their CALLERIDNUM and
prepends it to the number they dialed.
I've done something a bit different:
I use the Asterisk DB to store, for each extension, an associated
country code and area code. I then have a specific context, within
which I handle the number procesing. Thus, any number without a 0 in
front, gets +(COUNTRYCODE)(AREACODE) prepended. Any number with just a
0 in front, gets the country code added (+(COUNTRYCODE)), and a number
with the full 00 is considered a full number with country code and area
code. My code for this is as follows:
[processNumber]
exten = _ZX.,1,Set(COUNTRYCODE=${DB(extInfoCC/${CALLERID(NUM)})})
exten = _ZX.,n,Set(AREACODE=${DB(extInfoAC/${CALLERID(NUM)})})
exten = _ZX.,n,NoOp(Changing number from ${EXTEN} to
00${COUNTRYCODE}${AREACODE}${EXTEN})
exten = _ZX.,n,Goto(processNumber,00${COUNTRYCODE}${AREACODE}${EXTEN},1)
exten = _0ZX.,1,Set(COUNTRYCODE=${DB(extInfoCC/${CALLERID(NUM)})})
exten = _0ZX.,n,NoOp(Changing number from ${EXTEN} to
00${COUNTRYCODE}${EXTEN:1})
exten = _0ZX.,n,Goto(processNumber,00${COUNTRYCODE}${EXTEN:1},1)
exten = _00ZX.,1,NoOp(RepeatDial/${CALLERID(NUM)}=${EXTEN})
exten = _00ZX.,n,Set(DB(RepeatDial/${CALLERID(NUM)})=${EXTEN})
exten = _00ZX.,n,Set(NCDIAL=${EXTEN})
exten = _00ZX.,n,Goto(enumLookup,1)
exten = enumLookup,1,NoOp(Performing ENUM Lookup)
exten = enumLookup,n,Set(TRANSFER_CONTEXT=${SIPPEER(${CALLERID(NUM)}:context)})
exten =
enumLookup,n,Set(TARGET=${ENUMLOOKUP(+${NCDIAL:2},sip,record=1,e164.org)})
exten = enumLookup,n,GotoIf($[${LEN(${TARGET})}=0]?noEnum)
exten = enumLookup,n,SetAccount(ENUM-Alt)
exten = enumLookup,n,Dial(SIP/${TARGET},20,T)
exten = enumLookup,n,Hangup
exten = enumLookup,n(noEnum),Goto(routeNumber,${NCDIAL},1)
[routeNumber]
...
You'll have to change this as appropriate to work with US number formats.
HTH,
Roshan
--
http://roshan.info
All true wisdom is found through yo-yos.
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RE: [Asterisk-Users] TE405P vs. SoundCard problem
It's a clone built on an Intel 865GBF. Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Thursday, April 27, 2006 8:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE405P vs. SoundCard problem Bob - what type of server/mobo are you using? Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Bob McDowell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 27, 2006 8:46 AM Subject: [Asterisk-Users] TE405P vs. SoundCard problem My Asterisk server believes that a Digium TE405P and sound are incompatible. Basically, no matter what else I do to the machine it terms of hardware, if the TE405P is installed, none of the playback/background/etc commands work. MOH works fine. So far, I have tried: 1) Seven different PCI soundcards with different chipsets. (Go go Computer Junk Store!) 2) Disabling each and every device offered by the motherboard, except the IDE and video. 3) Each possible PCI slot combination via trial and error. 4) Calling Digium, who while helpful, did not know how to solve the problem. '/proc/interrupts' usually likes to put the 'wct4xxp' on the same line as my 'Intel ICH5', but as of this moment that isn't the case. The sound card, by the way, exists for overhead paging purposes. My questions are: 1) Is there a document I should be aware of? 2) Has anyone else resolved this sort of problem before? 3) Do I need to scrap the sound card and use an FX(O) device? If so, how do I get my sound back? Thanks, Bob McDowell *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. ___ --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] asterisk spandsp and txfax
Hello folks! I'm trying yo set up a email2fax and fax2email on my asterisk box. The rxfax works fine in my setup. The problem is with the txfax. I have tryed all snadsp version (0.0.2x and 0.0.3x) but I get this errors. Because I can't find anything on Internet I'm hoping u can give me a hand. here are my logs: -- Attempting call on SIP/sip_provider/1234 for application txfax(/va r/spool/asterisk/fax/215690048.1145968036.383.tif|caller|debug) (Retry 1) Channel SIP/mc3810-a20f was answered. Launching txfax(/var/spool/asterisk/fax/215690048.1145968036.383.tif|ca ller|debug) on SIP/mc3810-a20f FLOW Slow carrier up FLOW Slow carrier down FLOW Slow carrier up FLOW NSF: 20 00 00 0e 00 00 00 96 0f 41 07 00 10 00 02 95 80 18 01 49 02 53 2e 43 2e 4e 45 54 4d 41 53 54 45 52 20 20 20 03 FLOW NSF without final frame tag FLOW The remote was made by 'Panasonic' FLOW CSI: 40 38 30 30 30 39 36 35 31 32 30 20 20 20 20 20 20 20 20 20 20 FLOW CSI without final frame tag FLOW Remote fax gave CSI as: 0215690008 FLOW DIS: 80 00 ce 88 c4 80 11 FLOW DIS with final frame tag FLOW In state 10 FLOW ???: FLOW 3rd generation mobile network FLOW Prefer 256 octet blocks FLOW Reserved: 0x88 FLOW Supported data signalling rates: V.29 FLOW R8x7.7lines/mm and/or 200x200pels/25.4mm FLOW 2D coding FLOW Scan line length: 215mm FLOW Recording length: A4 (297mm) FLOW Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85 FLOW Reserved: 0x11 FLOW ???:FLOW Prefer 256 octet blocks FLOW Reserved: 0x80 FLOW Supported data signalling rates: V.27ter fallback mode FLOW 2D coding FLOW Scan line length: 215mm FLOW Recording length: A4 (297mm) FLOW Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85 FLOW Start sending document FLOW Changed from phase 2 to 4 FLOW DCS: 83 00 86 80 80 80 00 FLOW HDLC underflow in state 3 FLOW Changed from phase 4 to 6 FLOW Changed from phase 6 to 3 FLOW Slow carrier up FLOW Slow carrier down FLOW Slow carrier up FLOW NSF: 20 00 00 0e 00 00 00 96 0f 41 07 00 10 00 02 95 80 18 01 49 02 53 2e 43 2e 4e 45 54 4d 41 53 54 45 52 20 20 20 03 FLOW NSF without final frame tag FLOW The remote was made by 'Panasonic' FLOW T4 timeout in state 4 FLOW CSI: 40 38 30 30 30 39 36 35 31 32 30 20 20 20 20 20 20 20 20 20 20 FLOW CSI without final frame tag FLOW Remote fax gave CSI as: 0215690008 FLOW DIS: 80 00 ce 88 c4 80 11 FLOW DIS with final frame tag FLOW In state 4 FLOW Slow carrier down FLOW Slow carrier up FLOW NSF: 20 00 00 0e 00 00 00 96 0f 41 07 00 10 00 02 95 80 18 01 49 02 53 2e 43 2e 4e 45 54 4d 41 53 54 45 52 20 20 20 03 FLOW NSF without final frame tag FLOW The remote was made by 'Panasonic' FLOW CSI: 40 38 30 30 30 39 36 35 31 32 30 20 20 20 20 20 20 20 20 20 20 FLOW CSI without final frame tag FLOW Remote fax gave CSI as: 0215690008 FLOW DIS: 80 00 ce 88 c4 80 11 FLOW DIS with final frame tag FLOW In state 4 FLOW Slow carrier down FLOW Slow carrier up FLOW NSF: 20 00 00 0e 00 00 00 96 0f 41 07 00 10 00 02 95 80 18 01 49 02 53 2e 43 2e 4e 45 54 4d 41 53 54 45 52 20 20 20 03 FLOW NSF without final frame tag FLOW The remote was made by 'Panasonic' FLOW CSI: 40 38 30 30 30 39 36 35 31 32 30 20 20 20 20 20 20 20 20 20 20 FLOW CSI without final frame tag FLOW Remote fax gave CSI as: 0215690008 FLOW DIS: 80 00 ce 88 c4 80 11 FLOW DIS with final frame tag FLOW In state 4 FLOW Slow carrier down FLOW Slow carrier up FLOW NSF: 20 00 00 0e 00 00 00 96 0f 41 07 00 10 00 02 95 80 18 01 49 02 53 2e 43 2e 4e 45 54 4d 41 53 54 45 52 20 20 20 03 FLOW NSF without final frame tag FLOW The remote was made by 'Panasonic' FLOW CSI: 40 38 30 30 30 39 36 35 31 32 30 20 20 20 20 20 20 20 20 20 20 FLOW CSI without final frame tag FLOW Remote fax gave CSI as: 0215690008 FLOW DIS: 80 00 ce 88 c4 80 11 FLOW DIS with final frame tag FLOW In state 4 FLOW Slow carrier down FLOW Slow carrier up FLOW XCN: fa FLOW XCN with final frame tag FLOW In state 4 FLOW Disconnecting FLOW Changed from phase 3 to 7 FLOW Changed from phase 7 to 8 Apr 27 12:05:08 NOTICE[25889]: pbx_spool.c:279 attempt_thread: Call completed to SIP/sip_provider/1234 Hope u can give me a hint. Regards, Catalin Sarafoleanu. ___ --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] Need help configuring Asterisk with Alepo
HI I am trying to establish a connection between ASTERISK and ALEPO but I can not, since you have reached to make them communicate can you help me with the changes made to asterisk, in this way I will be able to check if the problem is the same with my ALEPO . I would appreciate every help you can give. Best Regards Dimal Telcom ___ --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] Interesting Dial-Plan Question
Ok that works... and I could do that if all I cared about was added
the 1 or country code. I guess in theory I could set a variable
set(ARECODE=${callerid(num)0:3}
_ = do stuff here for 7 digits
And then transform the number by taking the areacode and putting it in
front of the number, eh?
I guess I may just need to do it in contexts... right now I already
have a XXX that processes the call... maybe I just need to make a
context that sets the number up and then hand it off to the
dial-command context... ahh k... sometimes you just need to talk
things out and get some insight :)
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Re: [Asterisk-Users] Unable to accept incoming PSTN calls
[from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) Here is what is happening : Your ZAP channels are in the context from-pstn Since there is no s extension defined, it goes to _. (which match anything) So, like seen in the log, Asterisk wait a second, then execute Goto(from-pstr,s,1) which brings it back to _.,1. It just loop there until the caller hangup Since you're using [EMAIL PROTECTED], you have to go into AMP (or FreePBX) and click on Setup - Incoming Calls and define something to do with incoming calls hth ___ --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] Interesting Dial-Plan Question
exten = _NXX,1,Goto(${CALLERIDNUM::0:3}${EXTEN},1)
Matt wrote:
Eric,
Yes.. I am setting calleridnum to be their phone number. And your
example is peachy... except for the fact that it assumes I want to go
out ZAP/g1!!
My problem is I have a very intricite routing plan that routes that
call out several different carriers depending on what you dialed.
(Long Distance, international, local, etc).
The way it works now is the dialplan just looks at the number you
dialed and routes based on that. I guess what I am asking is in
theory I should be able to do:
Look at origination number. Take first 3 digits and put into
variable. So 5705551212 becomes 570 in ${AREACODE}.
Now, look at the number we dialed. If it is (and this is where I am a
little unclear on what to do) 7 digits long then we append the
${AREACODE} variable. Else, we send it through to the dialplan as
is.
exten = _NXX,1,Dial(Zap/g1/${CALLERIDNUM::0:3}${EXTEN})
This assumes that you set the user's Caller*ID number to be their
telephone number. It takes the first 3 digits of their CALLERIDNUM and
prepends it to the number they dialed.
See README.variables.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
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RE: [Asterisk-Users] Codec G729 / x86_64 bits.
At $10.00US per concurrent channel, it is better to buy, than to complain. Do you complain i someone gives you a new car but you have to pay for the gas?? (Bad example with Oil prices going high, but you get the point) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, April 27, 2006 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Codec G729 / x86_64 bits. how much are the codecs thst you cant buy em ? i dont intend to play judge and jurry however asterisk is a present that was given to all of us. im some way or another we should give to those that gave us. --- Jefferson Carvalho [EMAIL PROTECTED] wrote: Thanks for the suggestion , But I post a message to get a FREE codec (OPEN) , and not a PURCHASED. If I was interested in get a licensed one , believe that I never had Post a message on this list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Sunday, April 23, 2006 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Codec G729 / x86_64 bits. Jefferson Carvalho wrote: I always used a compiled version for a x86 system From [...] Someone could help me on this? Yes, the folks at Digium will be more than happy to help you. Visit http://www.digium.com/en/products/voice/g729codec.php and get a licensed codec. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Seize phone line
I have a question, we have some locations were I'm just planning on putting in a PRI, management also wants analog lines incase the PRI is down and someone calls 911. Is there a way to use asterisk to seize a phone line from the fax machine? I don't want to have to have an analog line that only gets used in the very rare situation with the PRI being down and someone needed to dial 911 (other incoming and outgoing calls would be routed over a private T1 to another location), but I don't want to just tap into the fax line because there is a chance that someone could be sending or receiving a fax at the same time. I found this: http://www.twacomm.com/catalog/model_LSR-1.htm on an internet search, anyone have any experience with this (or something similiar)? Would it work with asterik? On a related issue, at locations where we have 3 or 4 phone lines connected to asterisk and they are all in useand someone dials 911 we want it to disconnect one of the active calls so the 911 call can be made. Does anyone know how to do this? Would I need to use a device like the above or is there a way in software to do this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interesting Dial-Plan Question
That will work? So if I have:
CALLERIDNUM = 5705551212
exten = _NXX,1,Goto(${CALLERIDNUM::0:3}${EXTEN},1)
exten = _570NXX,1,Dial(Zap/g1/${EXTEN},1)
And if CALLERIDNUM = 7175551212
exten = _717NXX,1,Dial(Zap/g2/${EXTEN},1)
(Notice 717 calls go out g2.. and 570 go out g1).
That seems as though it should work, however it still would seem I
have to dork up my current dial-plan. That's why I'm wondering if I
should do that EXTEN and CALLERIDNUM stuff in another context, and
then transfer to my outdial context?
The problem I see doing it this way is that not ALL area code 717 or
570 calls go out g1 or g2. Some calls in the same are code are long
distance and need to route out the LD provider... even though they are
still 717-555-1212 format.
Right now.. if someone dials 1-570-555-1212, 570-555-1212, or 555-1212
it routes correctly (wether it is long distance or not). My goal is
to artificially make the person dial '570' or '717' even if they dial
555-1212 by looking at the originating number. Unless someone thinks
of a good reason not too.. it seems a context flip woudl be the way to
go.
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Re: [Asterisk-Users] Camp on?
On Thursday 27 April 2006 07:52, Rich Adamson wrote: Not likely either form can truly be implemented in a agi script without significantly impacting other pbx functions. I dunno... off the top of my head: - improve upon the standard extension macro such that any Dial() uses 'g' option, and include an 'h' option in the context - the macro will, upon Dial returning busy and detection of a local extension calling (done with variables) present a press 5 to receive a callback when this user is free, or 1 to go to voicemail or something similar. If a callback's required, save the extension of the calling party to a DB and hang up (I'm describing non-staying-on-the-line camp-on). - the macro will, upon the extension hanging up (using 'g' option), check the DB for a list of extensions waiting for callback and using a callout file, call the first one. - the sip/zap/iax extension's normal context's 'h' extension will do the same. - some housekeeping is required to keep the db clean but that's about it. that's your basic camp-on. while-you-wait campon isn't too different, and camp-on that calls you when they are around (i.e. they didn't answer but you want to know when they're back) is done similarly but when the monitored extension makes a call (done in the standard extension macro) or takes a call that is answered (standard extension looking at HANGUPCAUSE), it then recognizes that the person at the extension is there and the camp-on stuff above is activated. Yes this is all highlevel hand-waving at this point but I don't see where it's really impacting the PBX all that much. This can be done almost exclusively in the dialplan, without any AGIs or additional apps. Maybe a cron job to do the housekeeping and a script ot write out the proper callfiles... but that's it I think. -A. ___ --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] PRI configuration
Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 This is my zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=demo ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; qsig: Q.SIG ; switchtype=national ; ; Some switches (ATT especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'accunet' ; ;nsf=none ; group = 1 switchtype=national signalling=pri_cpe context=demo channel = 1-23 group = 2 switchtype=national signalling=pri_cpe context=demo channel = 25-47 Everything else is commented out, and I don't want to include them here. ___ --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] Seize phone line
Joe Pukepail wrote: I have a question, we have some locations were I'm just planning on putting in a PRI, management also wants analog lines incase the PRI is down and someone calls 911. Is there a way to use asterisk to seize a phone line from the fax machine? Multiple ways to do that. Something like the SPA3000 provides both an analog pstn interface and fxs interface (for the fax machine), and both of those interfaces are addressable via asterisk's dialplan. Or, use the sangoma A200D card with an fxo and fxs interface and you'll get the same functions (but with better quality). I don't want to have to have an analog line that only gets used in the very rare situation with the PRI being down and someone needed to dial 911 (other incoming and outgoing calls would be routed over a private T1 to another location), but I don't want to just tap into the fax line because there is a chance that someone could be sending or receiving a fax at the same time. I found this: http://www.twacomm.com/catalog/model_LSR-1.htm on an internet search, anyone have any experience with this (or something similiar)? Would it work with asterik? On a related issue, at locations where we have 3 or 4 phone lines connected to asterisk and they are all in use and someone dials 911 we want it to disconnect one of the active calls so the 911 call can be made. Does anyone know how to do this? Would I need to use a device like the above or is there a way in software to do this? Can't answer the above as it depends 100% on the exact equipment that you deploy to provide such services. Example, if you deployed the SPA300 (assuming analog phones) and you had it configured so as incoming pstn calls were directly connected to the fxs phones, asterisk has no control over the spa3000 path and can not dump existing calls. ___ --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] chan_sip.c patched with t.38
Hello, Is there Somebody to provide me a DID numder on a voip gateway which one support t.38 to test FOIP ? Regards Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --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] Guest Account - SIP and IAX
Dearest List, I understand how to handle guest calls via SIP and IAX. However, when such a call is placed, it will not look like IAX/guest-1234, or SIP/guest-1234. Instead, it will be something like IAX/the.callers.ip.address-1234 My issue is with getting this to map to a Flash Operator Panel button. I know regexp buttons can be made, even to regexp to an IP address. My question: is there a way to cause * to set any guest calls to the channel name like IAX2/guest-1234? Or is it better to just make a regexp button to match any valid IP address - this assumes that in * a guest call will always be formated as an IP, which I am not sure about... Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to accept incoming PSTN calls
Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not seem to be a menu area/settings for Incoming Calls? If you have a similiar setup, or know what the settings should be, could you possibly post them? If I were to create a dial group to ring all extensions, could that be used in place of s? Thanks kindly -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] Sent: Thursday, April 27, 2006 6:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) Here is what is happening : Your ZAP channels are in the context from-pstn Since there is no s extension defined, it goes to _. (which match anything) So, like seen in the log, Asterisk wait a second, then execute Goto(from-pstr,s,1) which brings it back to _.,1. It just loop there until the caller hangup Since you're using [EMAIL PROTECTED], you have to go into AMP (or FreePBX) and click on Setup - Incoming Calls and define something to do with incoming calls hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to accept incoming PSTN calls
For instance, I have tried the 2 below, but still it does not ring an existing extension, although the logs show it trying [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,SIP/100,1) or [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,100,1) -Original Message- From: Johnny Stork Sent: Thursday, April 27, 2006 7:11 AM To: asterisk-users Subject: RE: [Asterisk-Users] Unable to accept incoming PSTN calls Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not seem to be a menu area/settings for Incoming Calls? If you have a similiar setup, or know what the settings should be, could you possibly post them? If I were to create a dial group to ring all extensions, could that be used in place of s? Thanks kindly -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] Sent: Thursday, April 27, 2006 6:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) Here is what is happening : Your ZAP channels are in the context from-pstn Since there is no s extension defined, it goes to _. (which match anything) So, like seen in the log, Asterisk wait a second, then execute Goto(from-pstr,s,1) which brings it back to _.,1. It just loop there until the caller hangup Since you're using [EMAIL PROTECTED], you have to go into AMP (or FreePBX) and click on Setup - Incoming Calls and define something to do with incoming calls hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to accept incoming PSTN calls
Johnny,You need to setup an Inbound Route that matches all DIDs and all CIDs. In FreePBX, click on Inbound Routes, create a new route with blank CID and DID, and point it where you want it to go. It should work after that. AlexOn 4/27/06, Johnny Stork [EMAIL PROTECTED] wrote: Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not seem to be a menu area/settings for Incoming Calls?If you have a similiar setup, or know what the settings should be, could you possibly post them? If I were to create a dial group to ring all extensions, could that be used in place of s?Thanks kindly -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] ] Sent: Thursday, April 27, 2006 6:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) Here is what is happening : Your ZAP channels are in the context from-pstn Since there is no s extension defined, it goes to _. (which match anything) So, like seen in the log, Asterisk wait a second, then execute Goto(from-pstr,s,1) which brings it back to _.,1. It just loop there until the caller hangup Since you're using [EMAIL PROTECTED], you have to go into AMP (or FreePBX) and click on Setup - Incoming Calls and define something to do with incoming calls hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --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] Seize phone line
On a related issue, at locations where we have 3 or 4 phone lines connected to asterisk and they are all in use and someone dials 911 we want it to disconnect one of the active calls so the 911 call can be made. Does anyone know how to do this? Would I need to use a device like the above or is there a way in software to do this? In your dialplan where you handle 911 calls, you could just hangup a line (ex.: ZAP/4) then dial 911 on that line. If the line is not used, the hangup won't do anything, so no harm done. hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI configuration
Quoting http://www.asteriskguru.com/tutorials/e1t1.html -- configuration on SBC. If you are being flooded (several times a second, non stop and the pri never worked) by lines as: Jul 14 13:55:21 NOTICE[19519]: chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Then probably the PRI you are using is not using PRI signalling but maybe some other type of signalling like EM. On Thu, 2006-04-27 at 14:58, Wai Wu wrote: Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 This is my zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=demo ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; qsig: Q.SIG ; switchtype=national ; ; Some switches (ATT especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'accunet' ; ;nsf=none ; group = 1 switchtype=national signalling=pri_cpe context=demo channel = 1-23 group = 2 switchtype=national signalling=pri_cpe context=demo channel = 25-47 Everything else is commented out, and I don't want to include them here. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to accept incoming PSTN calls
[from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,100,1) Try somethin like [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) exten = s,1,Answer exten = s,2,Dial(SIP/100,20) hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI configuration
Also If you see the error Jul 14 13:55:21 NOTICE[19519]: chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 only occasionally, then you might have some devices in your pc (ide cards?) taking to long when taking an intterupt. You might want to try to put the te411p card on a different cpu, or if its probably an ide card doing it, try playing with hdparm (make your drivers slower) or disable that card, and take a new one. On Thu, 2006-04-27 at 14:58, Wai Wu wrote: Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 This is my zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=demo ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; qsig: Q.SIG ; switchtype=national ; ; Some switches (ATT especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'accunet' ; ;nsf=none ; group = 1 switchtype=national signalling=pri_cpe context=demo channel = 1-23 group = 2 switchtype=national signalling=pri_cpe context=demo channel = 25-47 Everything else is commented out, and I don't want to include them here. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Very stupid question regarding Polycom Soundstation 4000
Hi, we've had a couple of Sounstation 4000's around for a couple of months working fine with our * box. Today, I tried for the first time to do a local 3-way conference with one of them, and could not find the confrnc soft key for doing that (as stated in the user manual). Spent 20 minutes without success, and ended up transferring the other parties to a meetme conference. Anyone has one of those with a confrnc key? Thanks in advance, Cristian ___ --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] Extreme delay before * processes call files
Remco Barende wrote: Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call file just keeps sitting in the /var/spool/asterisk/outgoing directory and it seems that * is doing nothing with it?? Only after 10-30 seconds sometimes even much longer the call file is picked up. Are you *sure* you set up cron to run every second? I'm not aware of any official way to schedule cron-jobs more than once per minute... and that would probably explain what you're seeing. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI configuration
Thnks for the link. However, I know span 1 is pri because when I add the vpmsupport=0 parameter when loading wct4xxp, everything works and those messages don't show up. I think vpmsupport=0 parameter disable echo cancellation on the board (I have TE411P card). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Blades Sent: Thursday, April 27, 2006 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PRI configuration Quoting http://www.asteriskguru.com/tutorials/e1t1.html -- configuration on SBC. If you are being flooded (several times a second, non stop and the pri never worked) by lines as: Jul 14 13:55:21 NOTICE[19519]: chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Then probably the PRI you are using is not using PRI signalling but maybe some other type of signalling like EM. On Thu, 2006-04-27 at 14:58, Wai Wu wrote: Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 This is my zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=demo ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; qsig: Q.SIG ; switchtype=national ; ; Some switches (ATT especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'accunet' ; ;nsf=none ; group = 1 switchtype=national signalling=pri_cpe context=demo channel = 1-23 group = 2 switchtype=national signalling=pri_cpe context=demo channel = 25-47 Everything else is commented out, and I don't want to include them here. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interesting Dial-Plan Question
Ok here is what we did.. all in one context:
exten = _NXX,1,NoOp(Customer Area Code Is: ${CALLERIDNUM:0:3})
exten = _NXX,2,Goto(${CALLERIDNUM:0:3}${EXTEN},1)
exten = _1NX,1,NoOp(Chopping One Off Number: ${EXTEN:1:10})
exten = _1NX,2,Goto(${EXTEN:1:10},1)
Had to fix Eric's code a bit... CALLERIDNUM::0:3 didn't work :P But
that works great! Call lengths to US calls are now standardized :)
Before it showed up however you dialed it hehe.
On 4/27/06, Matt [EMAIL PROTECTED] wrote:
That will work? So if I have:
CALLERIDNUM = 5705551212
exten = _NXX,1,Goto(${CALLERIDNUM::0:3}${EXTEN},1)
exten = _570NXX,1,Dial(Zap/g1/${EXTEN},1)
And if CALLERIDNUM = 7175551212
exten = _717NXX,1,Dial(Zap/g2/${EXTEN},1)
(Notice 717 calls go out g2.. and 570 go out g1).
That seems as though it should work, however it still would seem I
have to dork up my current dial-plan. That's why I'm wondering if I
should do that EXTEN and CALLERIDNUM stuff in another context, and
then transfer to my outdial context?
The problem I see doing it this way is that not ALL area code 717 or
570 calls go out g1 or g2. Some calls in the same are code are long
distance and need to route out the LD provider... even though they are
still 717-555-1212 format.
Right now.. if someone dials 1-570-555-1212, 570-555-1212, or 555-1212
it routes correctly (wether it is long distance or not). My goal is
to artificially make the person dial '570' or '717' even if they dial
555-1212 by looking at the originating number. Unless someone thinks
of a good reason not too.. it seems a context flip woudl be the way to
go.
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RE: [Asterisk-Users] PRI configuration
I have a single scsi drive in the system. In a week or so, we will replace it with a sandisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gareth Blades Sent: Thursday, April 27, 2006 10:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PRI configuration Also If you see the error Jul 14 13:55:21 NOTICE[19519]: chan_zap.c:7874 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 only occasionally, then you might have some devices in your pc (ide cards?) taking to long when taking an intterupt. You might want to try to put the te411p card on a different cpu, or if its probably an ide card doing it, try playing with hdparm (make your drivers slower) or disable that card, and take a new one. On Thu, 2006-04-27 at 14:58, Wai Wu wrote: Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:24 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:26 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 This is my zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=demo ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; qsig: Q.SIG ; switchtype=national ; ; Some switches (ATT especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'accunet' ; ;nsf=none ; group = 1 switchtype=national signalling=pri_cpe context=demo channel = 1-23 group = 2 switchtype=national signalling=pri_cpe context=demo channel = 25-47 Everything else is commented out, and I don't want to include them here. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk as a phone survey system
I write perl applications for a living and have developed code to talk to all kinds of hardware. What I'd like to do is pull a list of phone numbers from sql via dbi and call each. An initial voice messsage would be played asking the recipient if they'd optionally like to fill out our survey. If so I'd like to on thefly play pre-recorded questions and record the touch tone response back into the database. Teleyapper looks like itdoes some of what I want but I'm not sure slicing it up is better than starting from scratch. There appear to be a fewCPAN modules to work with Asterisk. I'm looking for adviceon how hard this is to implement with Asterisk. TIA, TV JOEKerry Garrison [EMAIL PROTECTED] wrote: Asterisk is simply a telephony toolkit, so the simple answer is yes, Asterisk can do this. Also, being a toolkit means there are a number of ways to accomplish it. You could right PERL, Python, TCL, C, PHP or numerous other types of scripts that can manage this for you. To see how to do some of the basic functions, you can look at some of the scripts at Nerd Vittles (http://nerdvittles.com). Things like the TeleYapper will give you a basis to work from.Kerry GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net (949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.comFrom: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of TVJOESent: Wednesday, April 26, 2006 7:31 PMTo:asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asteriskas a phone survey system Hi,I'm interested in developing anautomated phone survey and am curious if Asterisk could beconfigured to run such a system.. My idea is to record a message anda series of sub-questions. The system would call each number on alist and play the message, Depending on thetouch tone responseanother message would be played. Is it possible for asterisk tomanage a survey like this? If so can the responses from thelisteners be recorded. If someone else has done this I'd beinterestedin details.TIA , TV JOE Yahoo!Messenger with Voice. PC-to-Phone calls for ridiculously low rates.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --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] Interesting Dial-Plan Question
Matt wrote:
That will work? So if I have:
CALLERIDNUM = 5705551212
exten = _NXX,1,Goto(${CALLERIDNUM::0:3}${EXTEN},1)
exten = _570NXX,1,Dial(Zap/g1/${EXTEN},1)
And if CALLERIDNUM = 7175551212
exten = _717NXX,1,Dial(Zap/g2/${EXTEN},1)
(Notice 717 calls go out g2.. and 570 go out g1).
That seems as though it should work, however it still would seem I
have to dork up my current dial-plan. That's why I'm wondering if I
should do that EXTEN and CALLERIDNUM stuff in another context, and
then transfer to my outdial context?
The problem I see doing it this way is that not ALL area code 717 or
570 calls go out g1 or g2. Some calls in the same are code are long
distance and need to route out the LD provider... even though they are
still 717-555-1212 format.
Right now.. if someone dials 1-570-555-1212, 570-555-1212, or 555-1212
it routes correctly (wether it is long distance or not). My goal is
to artificially make the person dial '570' or '717' even if they dial
555-1212 by looking at the originating number. Unless someone thinks
of a good reason not too.. it seems a context flip woudl be the way to
go.
You're on the right track. Do something like this:
[userdial]
exten = _NXXNXX,1,Goto(dial_us,${EXTEN},1)
exten = _NXX,1,Goto(dial_us,${CALLERIDNUM::0:3}${EXTEN},1)
[dial_us]
exten = _570NXX,1,Dial(Zap/g1/1${EXTEN},60)
exten = _717NXX,1,Dial(Zap/g2/1${EXTEN},60)
exten = _NXXNXX,1,Dial(Zap/g3/1${EXTEN},60)
Users will dial out of userdial. 7-digit numbers are prepended with
the first 3 digits of their caller-id, then sent to dial_us. 10-digit
numbers are sent there directly. dial_us chooses a trunk-group (or
IAX, SIP, whatever you want), based on the dialed area code(s).
A macro would probably be cleaner than a Goto, but either works in this
case.
FWIW, I'm a big fan of 1+10 dialing, as it removes some of the potential
ambiguity from the dial-plan.
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Re: [Asterisk-Users] Seize phone line
On 4/27/06, Rich Adamson [EMAIL PROTECTED] wrote: Joe Pukepail wrote: I have a question, we have some locations were I'm just planning on putting in a PRI, management also wants analog lines incase the PRI is down and someone calls 911.Is there a way to use asterisk to seize a phone line from the fax machine?Multiple ways to do that. Something like the SPA3000 provides both ananalog pstn interface and fxs interface (for the fax machine), and both of those interfaces are addressable via asterisk's dialplan. Or, use thesangoma A200D card with an fxo and fxs interface and you'll get the samefunctions (but with better quality). Aren't I asking for trouble by bridging fax traffic through asterisk? I have seen many reports on the mailing list that trying to fax through asterisk is problematic (at best) (until T.38 is implemented. ). ___ --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] Slip/Frame Error between Mitel SX-200 and Asterisk
I have a Dell PE SC420 (a no-no with a TE110P) connected to a Mitel SC-200. The Mitel gets Slip and Frame errors that cause the T1 card in the Mitel to go offline and this causes a service interruption. Could the SC-420/TE110P be causing these errors? I know it is listed on the incompatibility list, but do not know what side-effects are caused. Is this one of them ___ --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] some EICON Diva 4BRI questions
Hi Armin! Armin Schindler wrote: I'm not aware of such a cable to buy. Normaly, when you create a NT-side the connection is not made with just one cable (like I did because both device are just 10cm away from each other). In most cases you have an ISDN bus cabled in the rooms where the necessary changes (other termination, crossed) can be done in the boxes. I don't see any design bug here. Agreed - it's not a bug. Nevertheless having onboard terminators which canbe turned on/off with jumpers is a nice feature (like the quadBRI cards) regards klaus ___ --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] Seize phone line
Joe Pukepail wrote:
I have a question, we have some locations were I'm just planning on
putting in a PRI, management also wants analog lines incase the PRI is
down and someone calls 911. Is there a way to use asterisk to seize a
phone line from the fax machine?
I don't want to have to have an analog line that only gets used in
the very rare situation with the PRI being down and someone needed to
dial 911 (other incoming and outgoing calls would be routed over a
private T1 to another location), but I don't want to just tap into the
fax line because there is a chance that someone could be sending or
receiving a fax at the same time.
[dial911]
exten = 911,1,ChanIsAvail(${TRUNK_FAX})
exten = 911,2,Dial(${TRUNK_FAX}/911)
exten = 911,3,Hangup
exten = 911,102,SoftHangup(${TRUNK_FAX})
exten = 911,103,Wait(1)
exten = 911,104,Goto(1)
adapted from a wiki page.
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RE: [Asterisk-Users] Unable to accept incoming PSTN calls
I actually tried that before but it didnt seem to work. I tried once again and still nothing rings, whether I set the destination to a single extension, or a ring group. But the suggestion from another user below did work, but wont go to voicemail yet when its not answered. [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) exten = s,1,Answer exten = s,2,Dial(SIP/100,20) -Original Message-From: Alex Robar [mailto:[EMAIL PROTECTED]Sent: Thursday, April 27, 2006 7:32 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls Johnny,You need to setup an Inbound Route that matches all DIDs and all CIDs. In FreePBX, click on Inbound Routes, create a new route with blank CID and DID, and point it where you want it to go. It should work after that. Alex On 4/27/06, Johnny Stork [EMAIL PROTECTED] wrote: Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not seem to be a menu area/settings for "Incoming Calls"?If you have a similiar setup, or know what the settings should be, could you possibly post them? If I were to create a dial group to ring all extensions, could that be used in place of "s"?Thanks kindly -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] ] Sent: Thursday, April 27, 2006 6:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) Here is what is happening : Your ZAP channels are in the context "from-pstn" Since there is no "s" extension defined, it goes to "_." (which match anything) So, like seen in the log, Asterisk wait a second, then execute "Goto(from-pstr,s,1)" which brings it back to "_.,1". It just loop there until the caller hangup Since you're using [EMAIL PROTECTED], you have to go into AMP (or FreePBX) and click on Setup - Incoming Calls and define something to do with incoming calls hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --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] Excessive Asterisk delay to answer on ZAP inbound call
Hi Eric, this is my zapata.conf (zap/1 is a FXS but not used during tests): ;- ; Channel: zap/2 [in] - Telecom (lasciare libera) ;- language = us musiconhold = default signalling = fxs_ks channel = 2 usecallerid=no Note that omitting usecallerid=no does not change the result. Thanks to all for advices. Giorgio Incantalupo Eric ManxPower Wieling wrote: Paste your zapata.conf. Giorgio Incantalupo wrote: Hi Hadley, I tried usecallerid=no but unfortunately nothing changed. I used another pc with only one TDM400P because I thought I had too many TDM400P cards but I got the same behaviour. Giorgio Incantalupo Hadley Rich wrote: On Wednesday 26 April 2006 20:59, Giorgio Incantalupo wrote: Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? It's most likely waiting on callerid info. If you set usecallerid=no in your zapata.conf you should see it pick up faster, although without callerid. ___ --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 as a phone survey system
Could you kindly let us know what numbers those survey-calls will be coming from, so we can all add them to our blacklists? Thanks! TV JOE wrote: I write perl applications for a living and have developed code to talk to all kinds of hardware. What I'd like to do is pull a list of phone numbers from sql via dbi and call each. An initial voice messsage would be played asking the recipient if they'd optionally like to fill out our survey. If so I'd like to on the fly play pre-recorded questions and record the touch tone response back into the database. Teleyapper looks like it does some of what I want but I'm not sure slicing it up is better than starting from scratch. There appear to be a few CPAN modules to work with Asterisk. I'm looking for advice on how hard this is to implement with Asterisk. TIA, TV JOE ___ --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] access to caller/pickupgroup in extension.conf
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Is it possible to get the callergroup or pickupgroup of a phone in the dialplan? So I can make decisions depending on the caller/pickupgroup. chris... -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEUOdJR0exH8dhr/YRAggRAJ4vAskqbg+b4aI2GBM3/Kg7mNuQcwCgxLDo 8+BRKC55jJWLBGtsbpbC690= =FX2J -END PGP SIGNATURE- ___ --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] some EICON Diva 4BRI questions
On Thu, 27 Apr 2006, Klaus Darilion wrote: Hi Armin! Armin Schindler wrote: I'm not aware of such a cable to buy. Normaly, when you create a NT-side the connection is not made with just one cable (like I did because both device are just 10cm away from each other). In most cases you have an ISDN bus cabled in the rooms where the necessary changes (other termination, crossed) can be done in the boxes. I don't see any design bug here. Agreed - it's not a bug. Nevertheless having onboard terminators which canbe turned on/off with jumpers is a nice feature (like the quadBRI cards) Yeah, nice to have. But this is a setup thing only, so I won't call it a feature. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to accept incoming PSTN calls
I actually tried that before but it didnt seem to work. I tried once again and still nothing rings, whether I set the destination to a single extension, or a ring group. But the suggestion from another user below did work, but wont go to voicemail yet when its not answered. [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) exten = s,1,Answer exten = s,2,Dial(SIP/100,20) add this exten = s,3,Voicemail(u100) hth ___ --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
