Re: [asterisk-users] To use asterisk or proprietary hardware, that is the question
I would recommend you try the Escaux pbx.NET free edition for it. If the environment grows or commercial support is required they can upgrade or buy support from Escaux without any physical intervention (it's just a virtual flip switch on their systems). The GUI is webbased and quite simple. Apart from the administration on the website GUI, you can do a set and forget on the box, as it's debian based and can autoupdate itself. See http://www.escaux.com/index.php?option=com_contenttask=viewid=71Itemid=233lang=en (No I do not work for escaux, we've been testing it too here and I think it provides upgraded value for customers in need of support and understandable GUI) ___ --Bandwidth 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 and Cisco PRI gateway config
That was exactly what I meant. Your setup is : Nortel --- Cisco --- Asterisk What I was thinking about is: Nortel --- Asterisk1 --Asterisk2 In previous case, your are using Cisco's QSIG features. In the latter one, you could use Asterisk QSIG features. I was asking because, I was wondering how Cisco and Asterisk QSIG features compare. Sorry, I have no idea... __Yehavi: ___ --Bandwidth 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] Newbie would like some planning advice.
On Mon, 26 Feb 2007, Alan Chandler wrote: Ideally a DECT handset, which communicates with a basestation that either plugs into a LAN, or one of the computers would seem good and indeed the Philips VOIP1211 would seem to possibly fit the bill and be in a good price range. Unfortunately, all I can find out about its interface capability is that it is Skype compatible. The same is true of most other handsets that I can find - they say they are Skype compatible without giving any more detail. I'm playing with a pair of Siemens C460 DECT phones. The base stations have both a POTS line and a LAN connection. They aren't perfect in some respects, but are working very well. The Skype compatable ones will likely have a USB connection. Avoid if possible as you'll likely never get drivers to use the keypads on the phones with anything other than they version of Skype they provide with the phone. I was thinking of setting up Asterisk on my Linux Server and providing a limited service to my family as a PBX. But this will only be cost effective if I do not have to make more than a few pounds investment in handsets. The Siemens ones are more than a few pounds. (www.provu.co.uk) So you have a choice - use a soft phone (XLite, idefisk, etc.) with a USB handset, or headset (headset is preferable IMO, *IF* your PC has decent sound hardware) or spend the £££ on decent phones With that background, a few questions. 1.) If I keep everything at the SIP/RTP level, can I operate Asterisk on the server along with everything else. Its a 1.7G Celeron, and the loading from the other services is around 5% to 10%. Would be just fine for your applications. 2.) Can I make these so called Skype Compatible handsets work with Asterisk, or failing that are there any recommendations for alternatives which do work, but have the wireless capability described and don't cost a lot. Most of the ones I've seen are simple USB soundcard/microphone devices, nothing more. The hard part is driving the keys on the phones. You don't need to though as the soft phones (eg. xlite, idefisk) have keypads built in, so you use the mouse. Downside of soft phones is that you need the PC to be turned on and running the soft phone application... 3.) Can Asterisk manage the NAT traversal that will be necessary for communication from handsets on the WAN and handsets on the LAN. Port-forward 5060, 10,000 through 20,000 to the asterisk box inside your LAN, tell asterisk it's behind a NAT system (you need the appropriate runes of localnet, externip in the sip.conf file) and have the external phones use a STUN server. Gordon___ --Bandwidth 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] Dialling ZAP channel from analogue
On Sun, 25 Feb 2007 21:47:27 -0500, Jacob Helwig [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Take a look at the last line of your uri context. It looks like that is
matching before the outbound-local ones are. Try changing the extension
from _X. to _[0-8]X.
Looks like you're running into this issue:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
- --[ UxBoD ]-- wrote:
[outbound-local]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
[uri]
exten = _[a-z].,1,Macro(uridial,[EMAIL PROTECTED])
exten = _[A-Z].,1,Macro(uridial,[EMAIL PROTECTED])
exten = _X.,1,Macro(uridial,[EMAIL PROTECTED])
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFF4ko/RhLSniguQyERAmAnAKDBtpsmnjbB5/xKU4HelTV63JP7zACeKOf9
1ux2OKtkoCybfVQ38Pzn4ok=
=+LgZ
-END PGP SIGNATURE-
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--
This message has been scanned for viruses and dangerous content by
MailScanner, and is
believed to be clean.
I commented out the last line of the uri and that worked a treat. Though what
happens if I need to dial [EMAIL PROTECTED] ? This will not get matched, so is
there any way to say that if the domain is different from NULL then use the uri
context ?
--
--[ UxBoD ]--
// PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import
// Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8
// Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8
--
This message has been scanned for viruses and dangerous content by MailScanner,
and is
believed to be clean.
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[asterisk-users] Asterisk TE110P Hipath 3750
Has anyone go the above to work. Im trying to connect a siemens hipath 3750 to an asterisk server using a digium TE110P I can dial between the siemens to asterisk and vice versa, we just cannot dial out the main Line through the siemens pri. Any ideas. Regards Calvyn Disclaimer The information contained in this email is confidential and may contain proprietary information. It is meant solely for the intended recipient. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted in reliance on this, is prohibited and may be unlawful. No liability or responsibility is accepted if information or data is, for whatever reason corrupted or does not reach its intended recipient. No warranty is given that this email is free of viruses. The views expressed in this email are, unless otherwise stated, those of the author and not those of Prescient Investment Management or its management. Prescient Investment Management reserves the right to monitor, intercept and block emails addressed to its users or take any other action in accordance with its email use policy. Prescient Investment Management is an authorised Financial Services Provider. FSP No: 612 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Email From the dialplan
Hi Al,
We have different email contents and subject depending in the action that
the customer had taken, since our service is a Prepaid card. What I did, was
to use an AGI to have this flexibility. After getting the caller id and
querying it in the DB, I will then construct the email that will be sent to
the customer.
[nihonggo-temporary-registration]
exten = readnum,1,Set(REGPHONE=${CALLERIDNUM})
exten = 2,1,AGI(preRegist.agi,${REGPHONE})
exten = 2,2,AGI(sendEMAILnotification.agi,${REGPHONE})
AGI script..
sub mailSend {
my ($subj, $body) = @_;
my(@da, @day, @mon, $datetime, $expTO, $mail, $head);
@da = localtime(time);
@day = qw(Sun Mon Tue Wed Thu Fri Sat);
@mon = qw(Jan Feb Mar Apr May Jun Jul Aug Sep Oct Nov Dec);
$datetime = sprintf %s, %02d %s %d %02d:%02d:%02d +0900,
($day[$da[6]], $da[3]*1, $mon[$da[4]], $da[5]+1900,
$da[2]*1, $da[1]*1, $da[0]*1);
$body = Jcode-new($body)-h2z-jis;
$head = Return-Path: $m_RETPATH\n;
$expTO = join(,\n, @m_MAILTO);
$head .= To: $expTO\n;
$head .= From: $m_FROM\n;
$head .= Date: $datetime\n;
$head .= Content-type: text/plain; charset=iso-2022-jp\n;
$head .= Subject: $subj\n;
$head .= \n;
$head = mimeencode($head);
$mail = $head$body;
if (open (OUT, |$QIPATH)) { # qmail-inject
$mail =~ s/%/%%/g;
printf OUT $mail;
close (OUT);
}
}
Hope that helps.
Best Regards,
Joanna
www.mariazeta.com
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[asterisk-users] SetCIDNum is not available on 1.4svn
Hello, I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it on 1.4svn 56126 and it does not recognise this application. Any idea?... Thanks! __Yehavi: ___ --Bandwidth 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] SetCIDNum is not available on 1.4svn
Quoting Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED]: Hello, I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it on 1.4svn 56126 and it does not recognise this application. Any idea?... That application was deprecated in 1.2. To replace it, use: Set(CALLERID(num)=number) -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth 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] SetCIDNum is not available on 1.4svn
Yehavi Bourvine +972-8-9489444 wrote: I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it on 1.4svn 56126 and it does not recognise this application. Any idea?... SetCIDNum is deprecated since 1.2 (?). Use Set(CALLERID(num)=123) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth 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] IAX/SIP Inter Asterisk Transfer
Hi, I've thoroughly searched the mailing list for a concrete example but haven't found one, thats why I end up here with my question. Theres has been some talk about IAX Transfers, but quite theoretically. Lets say there is a setup of several asterisk-boxes connected in a local Ethernet. Asterisk A establishes actively a connection to a SIP client B. When B picks up, the connection should be transferred to another asterisk server C for further IVR-Stuff. The dialplan allows a Transfer = will this lead to a direct connection between B and C where I could take the network cable from A without losing the connection between B and C? B (SIP-Client)(SIP)--A(Asterisk) Asterisk A tries to connect to A by an Originate B picks Up there is a connection established B (SIP-Client)(SIP)-A(Asterisk)-(IAX|SIP??)-C(Asterisk) Asterisk A tries to transfer the SIP connection to Aseterisk C Will that lead to B-(SIP)---C (???) How would this look like in the dialplan? Lets say the originated Call from A to B ends up in context [Transfer_Context] A: [Transfer_Context] transfer = .. [Transfer_Context] extension = X,X,Transfer(Y) How fast would the transfer be if C picks up immediately, would B be able to recognise this? Thanks in advance, Knud ___ --Bandwidth 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] upgrading from A101 to....A102
Hi Jeremy, We had D channels problems with A102De (A102 with HWEC and PCI-Express version), and it was solved from Sangoma changing one parameter in wanpipe.conf. We have HP server too in this installation. Our problem with D-channel was when wanted use only half-E1 channels (really we continue having 15 channels up from telco), and we wanted limit them in wanpipe config. Here show you our wanpipe.conf: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 14 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES Our change was set ACTIVE_CH = ALL and every sync problems with telco about D-channels was solved. Hope this helps you. Regards On 23/2/07 17:16, Porier, Jeremy M. [EMAIL PROTECTED] wrote: We're having a lot of D channel problems with the pci-e on HP servers. Going to PCI fixed the problem. Sangoma is aware of the problem and is using one of our servers to work toward a solution. -Jeremy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Thursday, February 22, 2007 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] upgrading from A101 toA102 Any benefit on getting the PCI Express version? Bill ___ --Bandwidth 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] Out Proxy Call
Hello Users I have one VoIP service from Packet8 ( SIP protocal ) Packet8- Astetisk server - My SIP agents My Sip Agents are in Asterisk Server , I configured.. If any one user in My Asterisk has to Call the Packet8 service providers , How can I configure it. Till now I'm Doing on OpenSER and ASterisk (Voicemail and Confereing ..) But My Asterisk has to connect the Packet8 service providers please Help me... -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.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] How set CallerID via Macro or something
Hi guys, I need your help ... I have a couple of DIDs that reach my Asterisk box But I'd like to set my DIDs automatically via Macro or other routine based on the number called by my agent ... Ex: My agent called 954-111- ... So I'd like to set the callerid as 954-222- (That is my DID) Thanks in advance, Marcelo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How set CallerID via Macro or something
[EMAIL PROTECTED] wrote: Hi guys, I need your help ... I have a couple of DIDs that reach my Asterisk box But I'd like to set my DIDs automatically via Macro or other routine based on the number called by my agent ... Ex: My agent called 954-111- ... So I'd like to set the callerid as 954-222- (That is my DID) Thanks in advance, Marcelo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound caller ID or outbound? What kind of phones are you using? -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic 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] How set CallerID via Macro or something
Hi, I'm using traditional phones behing a Linksys PAP2 adapter ... I'd like to set the outbound caller ID ... based on the number dialed by my agent ... like ... If I dial one number with area code 781 and I have one DID with the same area code ... I'd like to set the caller ID to this number ... I tried already with a Macro but I didn't manage to retrieve a value from the Macro ... Thanks in advance, Marcelo -- Original message -- From: J. Oquendo [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 ___ --Bandwidth 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] AstriCon Europe 2007
Dear Asterisk Community, As some of you may know, we have held Asterisk conferences (AstriCons) in Europe the past two years. This year we are considering hosting another event, this time in Milan, Italy in late June. We would like to know how many members of the European Asterisk community would be interested in attending. The Milan event would last a total of two days and would include an Asterisk trade fair, tutorials, key-note addresses, industry perspectives and the ever popular Code Zone (a hacker's lounge). Topics would include both technical and business items. Tickets will sell for the same price as last year: 550 EUR. If you would be interested in attending as a delegate, as a speaker or as an exhibitor, please send an email to: [EMAIL PROTECTED] Let us know what you would like to see, if the location and dates work for you. We will be making a final decision later this week. Best Regards, Steve -- Steven Sokol CEO Sokol Associates, Inc. Asterisk Training: http://www.sokol-associates.com/ AstriCon 2007: http://www.astricon.net/ ___ --Bandwidth 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: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
Tzafrir all interested on it: After tried another options as CentOS 4.4 Server that didn't work, since I went back to hardware unstability as in FC4 I decide to retry FC6 x86_64. After install it, I run yum update yum, then yum update, then yum install kernel-devel. Then yum install kernel-devel-xen. This time I decide not to make the symlink to Linux26, since there was a lot of different opinions about if you need it or not. And then, I install zaptel-1.4.0 with ./configure, make menuselect, make all, make install make config. No more errors, it loads and works. Now I can start to play with the tdmoe and res_snmp, wich I couldn't get the xxx of the menu. Thanks, and I hope this time invested save someone else their time. Carlos Alperin ___ --Bandwidth 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] 1.4 res_snmp dependencies (Debian)
Did someone knows which are the dependencies on FC6? I get snmpd running but still I cannot load res_snmp module. Thanks, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, February 02, 2007 2:31 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 1.4 res_snmp dependencies (Debian) On Fri, Feb 02, 2007 at 12:49:26PM -0600, Jeremiah Millay wrote: I'm having trouble building res_snmp (under Asterisk 1.4.0) on a box running Debian Sarge. res_snmp says its dependencies are netsnmp but Debian doesn't seem to have a netsnmp package. I've tried installing pretty much every package available related to snmp and no luck. I'm just wondering if anyone has successfully built the res_snmp module under Debian Sarge stable. Any help or suggestions are appreciated. In Sarge: libsnmp5-dev In Etch: libsnmp9-dev In any case, 'apt-get install libsnmp-dev' would work. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How set CallerID via Macro or something
[EMAIL PROTECTED] wrote: Hi, I'm using traditional phones behing a Linksys PAP2 adapter ... I'd like to set the outbound caller ID ... based on the number dialed by my agent ... like ... If I dial one number with area code 781 and I have one DID with the same area code ... I'd like to set the caller ID to this number ... I tried already with a Macro but I didn't manage to retrieve a value from the Macro ... exten = _781NXX,1,Set(CALLERID(num)=7815551212) exten = _781NXX,n,Dial( exten = _NXXNXX,1,Set(CALLERID(num)=1235541313) exten = _NXXNXX,n,Dial( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How set CallerID via Macro or something
Thanks ... I think that is gonna work ... I don't how I didn't think about this ... Actually I think I was trying to do like through a generic way ... because I have a lot of DIDs I would have to set up one by one that way ... But thanks for your help ... But just to know .. Is there any way to retrieve any value from a macro ... ? Like ... I was trying to set the callerID dinamically inside the macro and set this value in the variable MACRO_RESULT ... but it didn't work ... Thanks, Marcelo -- Original message -- From: Eric ManxPower Wieling [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I'm using traditional phones behing a Linksys PAP2 adapter ... I'd like to set the outbound caller ID ... based on the number dialed by my agent ... like ... If I dial one number with area code 781 and I have one DID with the same area code ... I'd like to set the caller ID to this number ... I tried already with a Macro but I didn't manage to retrieve a value from the Macro ... exten = _781NXX,1,Set(CALLERID(num)=7815551212) exten = _781NXX,n,Dial( exten = _NXXNXX,1,Set(CALLERID(num)=1235541313) exten = _NXXNXX,n,Dial( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] 1.4 res_snmp dependencies (Debian)
On Mon, 2007-02-26 at 10:42 -0500, Carlos Alperin wrote: Did someone knows which are the dependencies on FC6? I get snmpd running but still I cannot load res_snmp module. Thanks, Maybe it does not load because it was not built correctly. For building it correctly you need to have these RPMs installed: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel 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] 1.4 res_snmp dependencies (Debian)
Thanks Patrick, I'll try Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Monday, February 26, 2007 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] 1.4 res_snmp dependencies (Debian) On Mon, 2007-02-26 at 10:42 -0500, Carlos Alperin wrote: Did someone knows which are the dependencies on FC6? I get snmpd running but still I cannot load res_snmp module. Thanks, Maybe it does not load because it was not built correctly. For building it correctly you need to have these RPMs installed: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel 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 ___ --Bandwidth 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 and Cisco PRI gateway config
Olivier wrote: That was exactly what I meant. Your setup is : Nortel --- Cisco --- Asterisk What I was thinking about is: Nortel --- Asterisk1 --Asterisk2 In previous case, your are using Cisco's QSIG features. In the latter one, you could use Asterisk QSIG features. I was asking because, I was wondering how Cisco and Asterisk QSIG features compare. I posted some questions about Q.SIG support in asterisk in the past here, actually I need caller id name transfer between siemens hipath pbx and sip phone connected to asterisk, I got some answers to my questions, even from one man from digium, says, that is _should_ work, but another here in formum post answers, that he can see caller id name from QSIG in asterisk, but in hex form, that asterisk can't decode, so, I think, that QSIG support in asterisk is currently not complete, especially not usefull for things like decoding caller id name from QSIG :'( PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How set CallerID via Macro or something
From: [EMAIL PROTECTED] Date: Mon, 26 Feb 2007 16:52:52 + Thanks ... I think that is gonna work ... I don't how I didn't think about this ... Actually I think I was trying to do like through a generic way ... because I have a lot of DIDs I would have to set up one by one that way ... But thanks for your help ... But just to know .. Is there any way to retrieve any value from a macro ... ? Like ... I was trying to set the callerID dinamically inside the macro and set this value in the variable MACRO_RESULT ... but it didn't work ... And wouldn't. There are some workarounds, including Gosub()..Return() structure. (Different from Macro but may work in some situations.) Yuan Liu Thanks, Marcelo -- Original message -- From: Eric ManxPower Wieling [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I'm using traditional phones behing a Linksys PAP2 adapter ... I'd like to set the outbound caller ID ... based on the number dialed by my agent ... like ... If I dial one number with area code 781 and I have one DID with the same area code ... I'd like to set the caller ID to this number ... I tried already with a Macro but I didn't manage to retrieve a value from the Macro ... exten = _781NXX,1,Set(CALLERID(num)=7815551212) exten = _781NXX,n,Dial( exten = _NXXNXX,1,Set(CALLERID(num)=1235541313) exten = _NXXNXX,n,Dial( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ex-Girlfriend syntax and RealTime Extensions
As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please give me one example. Regards, Ricardo. ___ --Bandwidth 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] Yellow or Red alarm on TE110P ????
i get this message with a red signal on TE110P card: * TE110P: span configured for... Calling startuo (flug is 4099) wcte1xxp: Setting yellow alarm * what does mean ? thank you :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions
Ricardo Carvalho wrote:
As seen in the following URL:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I
also tested some time ago with an old release of Asterisk, RealTime
Extensions didn't support the Ex-Girlfriend syntax.
Is it already working in recent 1.4 or 1.2.15 releases?
Not with dynamic Realtime.
Is there any other way that I can use to do the same thing but only
using contexts, for example? If yes, please give me one example.
You might use the Blacklist() application in 1.2 (deprecated!).
Using AstDB is an option:
exten = 123,1,Set(bad=${DB_EXISTS(exgirlfriends/${CALLERID(num)})})
exten = 123,n,GotoIf($[${bad} = 1]?blacklist,1)
exten = 123,n,Dial(SIP/${EXTEN})
exten = blacklist,1,Congestion()
exten = blacklist,n,Hangup()
Haven't checked if this works. You need to either set the DB
entries manually (see help database) or use DB() and
DB_DELETE() (might want to re-activate an ex-girlfriend).
Regards,
Philipp
--
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Let's use IT to solve problems and not to create new ones.
Asterisk - http://www.das-asterisk-buch.de
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Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions
Philipp Kempgen wrote: You might use the Blacklist() application in 1.2 (deprecated!). Sorry: LookupBlacklist() http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+LookupBlacklist has an example. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth 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] Call Forwarding
An odd question... I have asterisk running just basic sip phones and sending/receiving calls using ZAP. The phones are polycom 501s. When a user presses the Forward soft key and puts an external number (a cell phone), and then someone from the inside (another extension) to the phone which has the forward on... I get this odd and loud humming/buzz noise in place of what the ringer normally would be. Once the call completes, its fine. If you dial from the outside into the SIP phone, the forward happens just fine. Any thoughts? Rob ___ --Bandwidth 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] Ex-Girlfriend syntax and RealTime Extensions
Philipp Kempgen wrote: You might use the Blacklist() application in 1.2 (deprecated!). Using AstDB is an option: Is it really deprecated? I use LookupBlacklist (tested as working) in 1.4, and show application LookupBlacklist doesn't mention 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] Ex-Girlfriend syntax and RealTime Extensions
Thomas Kenyon wrote:
Philipp Kempgen wrote:
You might use the Blacklist() application in 1.2 (deprecated!).
Using AstDB is an option:
Is it really deprecated? I use LookupBlacklist (tested as working) in
1.4, and show application LookupBlacklist doesn't mention it.
But the source code (app_lookupblacklist.c) has this line:
ast_log(LOG_WARNING, LookupBlacklist is deprecated. Please use
${BLACKLIST()} instead.\n);
So I'm not sure if it's deprecated or not. But this points us
to a new function: BLACKLIST()
core show function BLACKLIST
---cut---
-= Info about function 'BLACKLIST' =-
[Syntax]
BLACKLIST()
[Synopsis]
Check if the callerid is on the blacklist
[Description]
Uses astdb to check if the Caller*ID is in family 'blacklist'. Returns
1 or 0.
---cut---
This just seems to fit.
Regards,
Philipp
--
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Let's use IT to solve problems and not to create new ones.
Asterisk - http://www.das-asterisk-buch.de
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[asterisk-users] Caller ID not getting to analog extensions
Hi Folks, Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this issue before (my other installations are always much larger than this one for home). -- Barry D. Hassler ___ --Bandwidth 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] Caller ID not getting to analog extensions
On Mon, Feb 26, 2007 at 04:31:07PM -0500, Barry D. Hassler wrote: Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this issue before (my other installations are always much larger than this one for home). Two hipshots: How *many* analog phones on your one FXS? And is it possible that the system is sending CNAME, not just CNID, and the phones don't do names, and are confused? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 ___ --Bandwidth 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] Caller ID not getting to analog extensions
There are 3 or 4 analog phones connected on the FXS port. Only 2 of them have callerID. On the CNAME as opposed to CNID, have NO idea! The callerID worked fine on these phones until I cut them over to the asterisk server this weekend. On 2/26/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Feb 26, 2007 at 04:31:07PM -0500, Barry D. Hassler wrote: Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this issue before (my other installations are always much larger than this one for home). Two hipshots: How *many* analog phones on your one FXS? And is it possible that the system is sending CNAME, not just CNID, and the phones don't do names, and are confused? Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Barry D. Hassler ___ --Bandwidth 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] Ex-Girlfriend syntax and RealTime Extensions
5. Set(BLACKLIST=${BLACKLIST()}) [pbx_config]
6. GotoIf($[${BLACKLIST} = 1]?shitlisted|s|1:7)
[pbx_config]
Philipp Kempgen wrote:
Thomas Kenyon wrote:
Philipp Kempgen wrote:
You might use the Blacklist() application in 1.2 (deprecated!).
Using AstDB is an option:
Is it really deprecated? I use LookupBlacklist (tested as working) in
1.4, and show application LookupBlacklist doesn't mention it.
But the source code (app_lookupblacklist.c) has this line:
ast_log(LOG_WARNING, LookupBlacklist is deprecated. Please use
${BLACKLIST()} instead.\n);
So I'm not sure if it's deprecated or not. But this points us
to a new function: BLACKLIST()
core show function BLACKLIST
---cut---
-= Info about function 'BLACKLIST' =-
[Syntax]
BLACKLIST()
[Synopsis]
Check if the callerid is on the blacklist
[Description]
Uses astdb to check if the Caller*ID is in family 'blacklist'. Returns
1 or 0.
---cut---
This just seems to fit.
Regards,
Philipp
Set(BLACKLIST=${BLACKLIST()})
GotoIf($[${BLACKLIST} = 1]?shitlisted|s|1:7)
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[asterisk-users] deprecated - CLI help vs. source code
Could someone with inside knowledge comment on that? If the
source code says deprecated but the CLI help does not mention
that - whom do I trust?
Original message
Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions
From: Philipp Kempgen [EMAIL PROTECTED]
Thomas Kenyon wrote:
Philipp Kempgen wrote:
You might use the Blacklist() application in 1.2 (deprecated!).
Using AstDB is an option:
Is it really deprecated? I use LookupBlacklist (tested as working) in
1.4, and show application LookupBlacklist doesn't mention it.
But the source code (app_lookupblacklist.c) has this line:
ast_log(LOG_WARNING, LookupBlacklist is deprecated. Please use
${BLACKLIST()} instead.\n);
So I'm not sure if it's deprecated or not. But this points us
to a new function: BLACKLIST()
[...]
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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RE: [asterisk-users] deprecated - CLI help vs. source code
Usage of LookupBlackList is deprecated. This means, the usage will work,
but there is no guarantee that it will work in future. You might want to
try using BLACKLIST() instead.
Thanks,
Neel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: Monday, February 26, 2007 3:07 PM
To: asterisk-dev@lists.digium.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] deprecated - CLI help vs. source code
Could someone with inside knowledge comment on that? If the
source code says deprecated but the CLI help does not mention
that - whom do I trust?
Original message
Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions
From: Philipp Kempgen [EMAIL PROTECTED]
Thomas Kenyon wrote:
Philipp Kempgen wrote:
You might use the Blacklist() application in 1.2 (deprecated!).
Using AstDB is an option:
Is it really deprecated? I use LookupBlacklist (tested as working) in
1.4, and show application LookupBlacklist doesn't mention it.
But the source code (app_lookupblacklist.c) has this line:
ast_log(LOG_WARNING, LookupBlacklist is deprecated. Please use
${BLACKLIST()} instead.\n);
So I'm not sure if it's deprecated or not. But this points us
to a new function: BLACKLIST()
[...]
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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RE: [asterisk-users] Yellow or Red alarm on TE110P ????
on February 26, 2007 12:49 PM younss azzayani said i get this message with a red signal on TE110P card: * TE110P: span configured for... Calling startuo (flug is 4099) wcte1xxp: Setting yellow alarm * what does mean ? I'm not sure about the middle part of this message but the yellow alarm part makes sense because if the card does not see a T1 signal coming in, a red alarm is set. When a piece of equipment goes into red alarm, it will automatically set the outgoing signal to a yellow alarm. A piece of equipment seeing this signal knows that the signal it is sending out is not arriving. The X represents a cable break. Equipment B is in red alarm Equipment B sends a yellow alarm back to equipment A Equipment A is in yellow alarm. |--| |--| | A|-X--| B | | | | | | || | |__| |__| Don Pobanz ___ --Bandwidth 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] Digium S101I echo - how to control it
Is there anyway to control echo on Digium S101I adapter? -- #Joseph ___ --Bandwidth 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] Digium S101I echo - how to control it
Joseph wrote: Is there anyway to control echo on Digium S101I adapter? Echo must be canceled at the VOIP/PSTN gateway. Because of the latencies involved in VoIP, canceling echo after a call has been converted to IP is not feasible. ___ --Bandwidth 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] To use asterisk or proprietary hardware, that is the question
On 2/24/07, shadowym [EMAIL PROTECTED] wrote: Astlinux would work except it does not currently meet some key requirements (GUI, Sangoma Analog card support). Otherwise it would be a GREAT distribution for set it and forget it running without a Hard Drive IMHO. shadowym, With a little work one could use the Digium GUI and the existing front end to rc.conf to provide a complete GUI for an AstLinux system using %100 open source components. Everything you need except the GUI itself is already there (PHP, http/https server - even sqlite). All you have to do is provide the PHP... As Darrick mentioned I have been working with Sangoma to get their cards supported in AstLinux again. Look out for the announcement of AstLinux 0.4.5 and AstLinux 0.5.0, both of which should feature full Sangoma support (along with Digium cards, of course)! Whether or not AstLinux can meet your needs it would be much better to use something that is OSS/Asterisk based. Looking at your requirements, it shouldn't be that hard. -- Kristian Kielhofner ___ --Bandwidth 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 Asterisk SIP Trunk and CallerID
I have two asterisk boxes (1.2.14) connected via SIP with Polycom 501s registered to each. I set callerid name and num before sending the call from one box to another but the phone registered to the receiving server only properly shows the caller name, not the number. The number on the phone always shows as the name of the sip registration of the calling server. Do I have to set a seperate sip header in the dialplan if I want to pass callerid name and number between two boxes? I feel like I'm making this too complicated. Thanks, Jeremy ___ --Bandwidth 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 Asterisk SIP Trunk and CallerID
Porier, Jeremy M. wrote: I have two asterisk boxes (1.2.14) connected via SIP with Polycom 501s registered to each. I set callerid name and num before sending the call from one box to another but the phone registered to the receiving server only properly shows the caller name, not the number. The number on the phone always shows as the name of the sip registration of the calling server. Do I have to set a seperate sip header in the dialplan if I want to pass callerid name and number between two boxes? I feel like I'm making this too complicated. Show us the line that sets the Caller*ID in your dialplan. ___ --Bandwidth 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] Ex-Girlfriend syntax and RealTime Extensions
Thanks for all those that replayed so far, Although I think your suggestions are interesting, I guess they don’t fit my needs, because I use the Ex-Girlfriend logic to find from which user each call comes in the PSTN direction, and selected the user by that way, calls are sent to the right DID. For example: exten = _[0-9]./tom,1,Dial(SIP/[EMAIL PROTECTED],120) exten = _[0-9]./john,1,Dial(SIP/[EMAIL PROTECTED],120) This kind of schema seems to don’t work with Dynamic Realtime, so how can I do the same thing using something that works with Dynamic Realtime? Thanks once again, Ricardo. ___ --Bandwidth 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 Asterisk SIP Trunk and CallerID
on the sending server I do this:
exten = s,3,Set(CALLERID(all)=My Name1234)
exten = s,4,Noop(${CALLERIDNAME})
exten = s,5,Noop(${CALLERIDNUM})
exten = s,6,Dial(SIP/to-ServerB/${MACRO_EXTEN})
for the record, it shows the correctly set callerid and name on 4 and 5. When
I do a Noop(${CALLERIDNUM}) on ServerB it shows fromServerA.
- Jeremy
From: [EMAIL PROTECTED] on behalf of Eric ManxPower Wieling
Sent: Mon 2/26/2007 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to Asterisk SIP Trunk and CallerID
Porier, Jeremy M. wrote:
I have two asterisk boxes (1.2.14) connected via SIP with Polycom 501s
registered to each. I set callerid name and num before sending the call
from one box to another but the phone registered to the receiving server
only properly shows the caller name, not the number. The number on the
phone always shows as the name of the sip registration of the calling
server.
Do I have to set a seperate sip header in the dialplan if I want to pass
callerid name and number between two boxes? I feel like I'm making this
too complicated.
Show us the line that sets the Caller*ID in your dialplan.
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Re: [asterisk-users] Asterisk to Asterisk SIP Trunk and CallerID
In your SIP config for servers A and B, do you specify callerid?
Porier, Jeremy M. wrote:
on the sending server I do this:
exten = s,3,Set(CALLERID(all)=My Name1234)
exten = s,4,Noop(${CALLERIDNAME})
exten = s,5,Noop(${CALLERIDNUM})
exten = s,6,Dial(SIP/to-ServerB/${MACRO_EXTEN})
for the record, it shows the correctly set callerid and name on 4 and 5. When I do a Noop(${CALLERIDNUM}) on ServerB it shows fromServerA.
- Jeremy
From: [EMAIL PROTECTED] on behalf of Eric ManxPower Wieling
Sent: Mon 2/26/2007 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to Asterisk SIP Trunk and CallerID
Porier, Jeremy M. wrote:
I have two asterisk boxes (1.2.14) connected via SIP with Polycom 501s
registered to each. I set callerid name and num before sending the call
from one box to another but the phone registered to the receiving server
only properly shows the caller name, not the number. The number on the
phone always shows as the name of the sip registration of the calling
server.
Do I have to set a seperate sip header in the dialplan if I want to pass
callerid name and number between two boxes? I feel like I'm making this
too complicated.
Show us the line that sets the Caller*ID in your dialplan.
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--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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[asterisk-users] Playback uses channel's language, background doesn't
I have the following in the dialplan:
[macro-systemrecording]
exten = s,1,Goto(${ARG1},1)
exten = dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav)
exten = dorecord,n,Wait(1)
exten = dorecord,n,Goto(confmenu,1)
exten = docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording)
exten = docheck,n,Wait(1)
exten = docheck,n,Goto(confmenu,1)
exten =
confmenu,1,Background(to-listen-to-itpress-1to-rerecord-itpress-star|m||macro-systemrecording)
exten = confmenu,n,Read(RECRESULT||1|||4)
exten = confmenu,n,GotoIf($[x${RECRESULT}=x*]?dorecord,1)
exten = confmenu,n,GotoIf($[x${RECRESULT}=x1]?docheck,1)
exten = confmenu,n,Goto(1)
exten = 1,1,Goto(docheck,1)
exten = *,1,Goto(dorecord,1)
exten = t,1,Playback(goodbye)
exten = t,n,Hangup
exten = i,1,Playback(pm-invalid-option)
exten = i,n,Goto(confmenu,1)
exten = h,1,Hangup
When this is called the following is shown in the CLI
-- Goto (macro-systemrecording,docheck,1)
-- Executing Playback(SIP/223344-0928bbb8, /tmp/2595-ivrrecording) in
new stack
-- Playing '/tmp/2595-ivrrecording' (language 'nz')
-- Executing Wait(SIP/223344-0928bbb8, 1) in new stack
-- Executing Goto(SIP/223344-0928bbb8, confmenu|1) in new stack
-- Goto (macro-systemrecording,confmenu,1)
-- Executing BackGround(SIP/223344-0928bbb8,
to-listen-to-itpress-1to-rerecord-itpress-star|m||macro-systemrecording)
in new stack
-- Playing 'to-listen-to-it' (language '')
As can be seen, Playback uses the channel's language 'nz' but BackGround does
not. Could anyone advise what I'm doing wrong?
Thanks
Cameron
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Re: [asterisk-users] Playback uses channel's language, background doesn't
it may be a bug, try creating a simple test script with only 2
extensions, one with playback the other one with background and see
how it works, also post here the asterisk version you are using.
Regards
On 2/26/07, kjcsb [EMAIL PROTECTED] wrote:
I have the following in the dialplan:
[macro-systemrecording]
exten = s,1,Goto(${ARG1},1)
exten =
dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav)
exten = dorecord,n,Wait(1)
exten = dorecord,n,Goto(confmenu,1)
exten =
docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording)
exten = docheck,n,Wait(1)
exten = docheck,n,Goto(confmenu,1)
exten =
confmenu,1,Background(to-listen-to-itpress-1to-rerecord-itpress-star|m||macro-systemrecording)
exten = confmenu,n,Read(RECRESULT||1|||4)
exten =
confmenu,n,GotoIf($[x${RECRESULT}=x*]?dorecord,1)
exten =
confmenu,n,GotoIf($[x${RECRESULT}=x1]?docheck,1)
exten = confmenu,n,Goto(1)
exten = 1,1,Goto(docheck,1)
exten = *,1,Goto(dorecord,1)
exten = t,1,Playback(goodbye)
exten = t,n,Hangup
exten = i,1,Playback(pm-invalid-option)
exten = i,n,Goto(confmenu,1)
exten = h,1,Hangup
When this is called the following is shown in the CLI
-- Goto (macro-systemrecording,docheck,1)
-- Executing Playback(SIP/223344-0928bbb8, /tmp/2595-ivrrecording)
in new stack
-- Playing '/tmp/2595-ivrrecording' (language 'nz')
-- Executing Wait(SIP/223344-0928bbb8, 1) in new stack
-- Executing Goto(SIP/223344-0928bbb8, confmenu|1) in new stack
-- Goto (macro-systemrecording,confmenu,1)
-- Executing BackGround(SIP/223344-0928bbb8,
to-listen-to-itpress-1to-rerecord-itpress-star|m||macro-systemrecording)
in new stack
-- Playing 'to-listen-to-it' (language '')
As can be seen, Playback uses the channel's language 'nz' but BackGround
does not. Could anyone advise what I'm doing wrong?
Thanks
Cameron
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[asterisk-users] Asterisk - Streaming Audio Bridge
Greetings, Does anyone know of a tool that can act as a VoIP client and stream to a streaming server such as shoutcast/icecast, etc. I've got a client interested in doing basketball play by plays during tourney season. They have * in place now and the bandwidth to burn for streaming out. In the old world, I did an analog phone patch - mixer - encoder - streaming server. What I'm thinking of is more along the lines of a client that registers as a SIP/IAX client, answers the phone and patches it to a streaming server. Thoughts/suggestions? Thanks Eric ___ --Bandwidth 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 - Streaming Audio Bridge
Have you looked at ICEcast? http://www.icecast.org/index.php - Original Message - From: Eric Germann [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, February 26, 2007 6:17 PM Subject: [asterisk-users] Asterisk - Streaming Audio Bridge Greetings, Does anyone know of a tool that can act as a VoIP client and stream to a streaming server such as shoutcast/icecast, etc. I've got a client interested in doing basketball play by plays during tourney season. They have * in place now and the bandwidth to burn for streaming out. In the old world, I did an analog phone patch - mixer - encoder - streaming server. What I'm thinking of is more along the lines of a client that registers as a SIP/IAX client, answers the phone and patches it to a streaming server. Thoughts/suggestions? Thanks Eric ___ --Bandwidth 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] XM Radio Stream to Asterisk
Hi Guys, Anyone figure out a way to have XM radio work over asterisk (Not thru an audio card but over the internet) ? Either where a sip phone dials an extension (i.e. *202 will go to XM channel 202) or maybe with a confrence room so multiple people can call in and listen. Thanks for any and all ideas. Dovid___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO
Hi José, I have not resolve this issue yet. I am currently focusing in my newly arrived toy (fonebridge2) then after which I will go back to AudioCodes Issue. Still I don't received yet any response from AudioCodes Representative here in the Philippines. I had already escalated this to their Regional Office in Singapore. But still no reply for almost a month already. I will post immediately once I resolve the issue. It is important to us because we really need to now where the calls coming from. Regards Angel. José Luis Gómez [EMAIL PROTECTED] wrote: Hello Angel. Did you solve this issue? I have the same problem. Thanks, José El jue, 01-02-2007 a las 01:15 -0800, Angel Heart escribió: Hi, I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming outgoing calls. However, I noticed that the caller ID of the caller coming from the FXO displays its endpoints assigned number and not the actual caller's ID coming from PSTN. Hope someone is using the same scenario and could share on how to resolve the caller ID/Number. Thanks. Angel __ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. ___ --Bandwidth 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 - Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.___ --Bandwidth 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] XM Radio Stream to Asterisk
Oh Dovid, You always seem to be up to something! All these strange projects ;). On 2/26/07, Dovid B [EMAIL PROTECTED] wrote: Hi Guys, Anyone figure out a way to have XM radio work over asterisk (Not thru an audio card but over the internet) ? Either where a sip phone dials an extension (i.e. *202 will go to XM channel 202) or maybe with a confrence room so multiple people can call in and listen. Thanks for any and all ideas. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] XM Radio Stream to Asterisk
I like my OA. If I am running around I dont wana miss a moment ;) - Original Message - From: mitcheloc [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 27, 2007 5:41 AM Subject: Re: [asterisk-users] XM Radio Stream to Asterisk Oh Dovid, You always seem to be up to something! All these strange projects ;). On 2/26/07, Dovid B [EMAIL PROTECTED] wrote: Hi Guys, Anyone figure out a way to have XM radio work over asterisk (Not thru an audio card but over the internet) ? Either where a sip phone dials an extension (i.e. *202 will go to XM channel 202) or maybe with a confrence room so multiple people can call in and listen. Thanks for any and all ideas. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] To use asterisk or proprietary hardware, that is the question
Thanks for the comments Kristian, I don't really have the skills to make the Digium GUI work on astlinux but if it is that natural of a fit I look forward to the day someone does it and makes it publicly available. Astlinux + Digium GUI + Sangoma analog card support would be a magical combination IMHO! -Original Message- From: Kristian Kielhofner [mailto:[EMAIL PROTECTED] Sent: Monday, February 26, 2007 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] To use asterisk or proprietary hardware,that is the question On 2/24/07, shadowym [EMAIL PROTECTED] wrote: Astlinux would work except it does not currently meet some key requirements (GUI, Sangoma Analog card support). Otherwise it would be a GREAT distribution for set it and forget it running without a Hard Drive IMHO. shadowym, With a little work one could use the Digium GUI and the existing front end to rc.conf to provide a complete GUI for an AstLinux system using %100 open source components. Everything you need except the GUI itself is already there (PHP, http/https server - even sqlite). All you have to do is provide the PHP... As Darrick mentioned I have been working with Sangoma to get their cards supported in AstLinux again. Look out for the announcement of AstLinux 0.4.5 and AstLinux 0.5.0, both of which should feature full Sangoma support (along with Digium cards, of course)! Whether or not AstLinux can meet your needs it would be much better to use something that is OSS/Asterisk based. Looking at your requirements, it shouldn't be that hard. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get values of local channels context
The variable ${CONTEXT} stores the value of the current context. However if we
are in a macro that will be the name of the macro. How do I access the name of
the local channel's context.
For example:
[macro-test]
exten = s,n,NoOp(Context ${CONTEXT})
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Context macro-test) in new
stack
I want to get 116-2000 somehow.
Any suggestions would be appreciated.
Cameron
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Re: [asterisk-users] Asterisk - Streaming Audio Bridge
On 26 Feb 2007, at 18:17, Eric Germann wrote: Greetings, Does anyone know of a tool that can act as a VoIP client and stream to a streaming server such as shoutcast/icecast, etc. I've got a client interested in doing basketball play by plays during tourney season. They have * in place now and the bandwidth to burn for streaming out. In the old world, I did an analog phone patch - mixer - encoder - streaming server. What I'm thinking of is more along the lines of a client that registers as a SIP/IAX client, answers the phone and patches it to a streaming server. Thoughts/suggestions? You could consider skipping the streaming server and just having the users hit asterisk directly - It depends on the numbers of users you want to serve, but you could drop them all into a muted app_conference. That way you get mush less latency than you do with a streaming solution, which might matter with something as quick as a basketball game. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth 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] Playback uses channel's language, background doesn't
it may be a bug, try creating a simple test script with only 2 extensions, one with playback the other one with background and see how it works, also post here the asterisk version you are using. Asterisk 1.2.13 exten = 98765,1,Playback(to-listen-to-it) exten = 98764,1,Background(to-listen-to-it|m||macro-systemrecording) exten = 98763,1,Background(to-listen-to-it) -- Executing Playback(SIP/112233-09289b40, to-listen-to-it) in new stack -- Playing 'to-listen-to-it' (language 'nz') -- Executing Hangup(SIP/112233-09289b40, ) in new stack == Spawn extension (116-2000, h, 1) exited non-zero on 'SIP/112233-09289b40' -- Executing BackGround(SIP/112233-09289b40, to-listen-to-it|m||macro-systemrecording) in new stack -- Playing 'to-listen-to-it' (language '') -- Executing Hangup(SIP/112233-09289b40, ) in new stack == Spawn extension (116-2000, h, 1) exited non-zero on 'SIP/112233-09289b40' -- Executing BackGround(SIP/112233-09289b40, to-listen-to-it) in new stack -- Playing 'to-listen-to-it' (language 'nz') -- Executing Hangup(SIP/112233-09289b40, ) in new stack == Spawn extension (116-2000, h, 1) exited non-zero on 'SIP/112233-09289b40' So it seems assume that since I passed a blank language override to the Background application, that I want a blank language. Any ideas on how to get background to use the default language? Regards Cameron ___ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO
I have no experience with AudioCodes but it seems that you need to have callerID enabled, leave endpoint phone number blank. Hope this helps. Maybe some of this info might help: http://www.voip-info.org/wiki/view/AudioCodes ++ If caller ID is turn on, then freePBX will only record the receiving number.not the line number. ++ Well, you can fix this, by using the Routing General settings, Audiocodes allows you to Prepend the Hunt Group to the number, You can then use the Manipulation tables, and strip the source number (tel--IP) after Routing. So if u set each Endpoint up to have a different Hunt Group, you can get it to ID the line. They also have a x-channel header that can be added for you to look in the SIP message at. things that help when dealing with the FXO's They are designed to work with Analog PSTN lines, 1. Caller ID is usually delivered between the 1st and 2nd ring on these lines. Also make sure it is enabled in the Supplementary services. 2. For those of you expecting the number to get delivered through to the IP side when dialing, it won't PBX's and CO's just ring the PSTN line, they don't deliver the number. Make sure you Enable AutoMatic Dialing in the Endpoint Settings, and if you want the line in port x to be the number dialed to the sip side, datafill the number there. 3. Make sure you set up the audiocodes with the proper coder like Ulaw, they come set to 723 by default which is crap for coders. they can support up to 5 so just datafill them with all the big coders U, A, 729, and whatever else 4. The Advanced Configuration pages have all their Channel settings, make sure the fax's are set to what the Trixbox supports. Audiocodes by default does t.38 now. if your pbx isn't set up for it, you need to put the Audiocodes in a transparent or events mode If you want the source number from IP to use the same datafilled Endpoint Port on the PSTN side make sure Endpoint Phone Numbers has that number datafilled, and then set up a hunt group with source number as the selection algorithm(5.0). Assign the endpoints to that hunt group. IP to Tel rouitng route all calls to that group Endpoint Phone Number - This will give you the options for either 4 or 8 ports. You do not need to place anything here. However, it is a good idea to do such to help you identify which port the call comes in on; as you can view the reports in freePBX to identify calls. In my case, since I have four PSTN ports, I used the last four digits of the telephone number to identify. Identifying which PSTN line the call came from only works if you DO NOT have caller id on the line, or your turn off caller id. If caller ID is turn on, then freePBX will only record the receiving number.not the line number. Endpoint Settings - Automatic Dialing - Define a station number located on Asterisk / Trixbox (ie 101) for all ports - Caller ID - Allowed .. turn off if you want to Identify the line they came in on. - Detect Caller ID from Tel - Enable Thanks, Steve Totaro From: Angel Heart [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO Date: Mon, 26 Feb 2007 19:22:53 -0800 (PST) Hi José, I have not resolve this issue yet. I am currently focusing in my newly arrived toy (fonebridge2) then after which I will go back to AudioCodes Issue. Still I don't received yet any response from AudioCodes Representative here in the Philippines. I had already escalated this to their Regional Office in Singapore. But still no reply for almost a month already. I will post immediately once I resolve the issue. It is important to us because we really need to now where the calls coming from. Regards Angel. José Luis Gómez [EMAIL PROTECTED] wrote: Hello Angel. Did you solve this issue? I have the same problem. Thanks, José El jue, 01-02-2007 a las 01:15 -0800, Angel Heart escribió: Hi, I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming outgoing calls. However, I noticed that the caller ID of the caller coming from the FXO displays its endpoints assigned number and not the actual caller's ID coming from PSTN. Hope someone is using the same scenario and could share on how to resolve the caller ID/Number. Thanks. Angel __ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] How to get values of local channels context
On Mon, Feb 26, 2007 at 08:06:40PM -0800, kjcsb wrote:
The variable ${CONTEXT} stores the value of the current context. However if
we are in a macro that will be the name of the macro. How do I access the
name of the local channel's context.
For example:
[macro-test]
exten = s,n,NoOp(Context ${CONTEXT})
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Context macro-test) in
new stack
I want to get 116-2000 somehow.
Any suggestions would be appreciated.
From the text of 'show application macro':
[Description]
Macro(macroname|arg1|arg2...): Executes a macro using the context
'macro-macroname', jumping to the 's' extension of that context and
executing each step, then returning when the steps end.
The calling extension, context, and priority are stored in ${MACRO_EXTEN},
${MACRO_CONTEXT} and ${MACRO_PRIORITY} respectively. Arguments become
${ARG1}, ${ARG2}, etc in the macro context.
So use ${MACRO_CONTEXT} .
--
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+972-50-7952406 mailto:[EMAIL PROTECTED]
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RE: [asterisk-users] To use asterisk or proprietary hardware, that is the question
Thanks Tom and everyone else, Based largely on your comments I decided to just stick with what works. I have a site using entry level ATX server hardware that has been solid as a rock. I'll just go with that instead of more specialized fanless hardware, specialized power supply and 2.5 hard drives etc. Maybe get a second motherboard as a spare of they go for the ongoing remote support option. I'll do some simple things like a put in a standby hard drive with the production image on it in case the primary drive fails. The case has hot swap SATA bays so if the primary drive fails or get's corrupted anyone can just swap drives and they will be back up just like that. I'll make remote offsite backups as well. Thanks for all the help. -Original Message- From: Tom [mailto:[EMAIL PROTECTED] Sent: Saturday, February 24, 2007 5:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] To use asterisk or proprietary hardware,that is the question At 11:53 AM 2/24/2007, you wrote: Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a set it and forget it type of thing like a lot of small business proprietary systems. There is no such thing as set and forget. Businesses change. They either grow or shrink, they don't stand still. They will add and remove phones. So they will call you at that time. Or are you expecting them to shop for their own phones on Ebay? At the same time they would like to be able to do minor dial plan changes themselves so I have determine that a GUI like FreePBX or similar alternative (free or commercial) is appropriate. We take a different approach. We don't want a GUI. We don't want the limits. We work with the business to design their dial plan. Then we write it. We do not give them a GUI because we don't want them making changes and then asking for support. We sell them a minor service agreement and remote in for any changes. We also handle professional voice recording and basic training on phone use. And we handle backups and service if needed. Once they understand that we can do that without a service call, they are quite receptive to the idea. Conventional PBXs come with service agreements so customers are used to that but surprised at the low cost from you. I have some concerns about using Asterisk for this. As much as I am in support of the whole Asterisk revolution, I just do not feel confident enough in Asterisk on a Hard Drive as a set it and forget it setup running month after month, year after year. I am hoping someone can convince me otherwise. Hard drives are reliable. But I have similar feelings so we are working on a flash solution. Were running it beta in our office right now. It only uses the hard drive for daily voicemail, boots from flash and runs from RAM. I'm concerned about hard drive corruptions/failures, memory leaks, software bugs etc. Conventional systems have bugs too. I have the budget to buy good quality hardware so if I was to go with Asterisk I would go industrial grade fanless computer, power conditioned UPS etc. You don't really need fanless. Make it cheap enough that it can easily be replaced. Like a $500 PC. I am not concerned about the reliability of most of the hardware. It's the hard drive and the software that runs on it that worries me. I will obviously use a mature stable Asterisk release and the most stable Linux version which I won't bother naming just to keep the discussion focussed. Asterisk is pretty darn stable. I have other Asterisk installs that went well but they were in environments where there were IT people around who were prepared to deal with some Linux administration and I could provide ongoing support for more major things. That is not the case here. Some of those sites have been running for months untouched, some needed some updates and reboots for various issues. I don't think this customer would look very favorably on me having to come in and add patches or have to reboot once a month or whatever. So do it from home. And how often do you really need to upgrade a minimal read only flash based system with no dev tools running from RAM? Does the latest kernel really matter? Their expection is the same as they would have with any other phone system that mounts on the wall and just works for years. I think that is a reasonable expectation. Agreed. And if it breaks, you replace it quickly and at a low cost. I am looking at putting in an Epygi proprietary VoIP system in instead. It is mostly hardware based although apparently runs Linux. It has a GUI, is supposedly plug and play most of the time, and most importantly, does not use a Hard Drive. I have heard good things about them so for arguments sake, let's assume voice quality, features, and the enduser experience are
Re: [asterisk-users] How to get values of local channels context
kjcsb wrote:
The variable ${CONTEXT} stores the value of the current context. However
if we are in a macro that will be the name of the macro. How do I access
the name of the local channel's context.
For example:
[macro-test]
exten = s,n,NoOp(Context ${CONTEXT})
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2
mailto:Local/[EMAIL PROTECTED],2, Context macro-test) in new stack
I want to get 116-2000 somehow.
Hi,
Check out ${MACRO_CONTEXT}
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro
--
Warm Regards,
Lee
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Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO
This sounds promising and obvious. Two rings is pretty standard for analog caller ID. I would like to add some additional insight into #7. Unless you have the MP-114 hanging off a PBX with caller ID, you should probably set the PROTOCOL MANAGEMENT FXO SETTINGS Rings before detecting caller ID to 1. Most Class 5 offices will not give you caller ID messaging info until sometime during the first cycle; as this info comes from the SS7 channel with lower priority than the actual signaling info. Whereas, if you are hanging off a PBX, by the time the PBX trunk recognizes a seizure, the caller ID info is delivered and the PBX makes cut-through to the station delivering both ring generator and caller ID as the same time. Setting it to 0 off a class 5 could either give you the caller ID content you defined in the station ID info in ENDPOINT fields, or if you put nothing in there, unknown caller, or 1000 (default setting PROTOCOL MANAGEMENT PROTOCOL DEFINITION DTMF DIALING as you defined in #5. From: Steven Totaro [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO Date: Mon, 26 Feb 2007 23:37:25 -0500 I have no experience with AudioCodes but it seems that you need to have callerID enabled, leave endpoint phone number blank. Hope this helps. Maybe some of this info might help: http://www.voip-info.org/wiki/view/AudioCodes ++ If caller ID is turn on, then freePBX will only record the receiving number.not the line number. ++ Well, you can fix this, by using the Routing General settings, Audiocodes allows you to Prepend the Hunt Group to the number, You can then use the Manipulation tables, and strip the source number (tel--IP) after Routing. So if u set each Endpoint up to have a different Hunt Group, you can get it to ID the line. They also have a x-channel header that can be added for you to look in the SIP message at. things that help when dealing with the FXO's They are designed to work with Analog PSTN lines, 1. Caller ID is usually delivered between the 1st and 2nd ring on these lines. Also make sure it is enabled in the Supplementary services. 2. For those of you expecting the number to get delivered through to the IP side when dialing, it won't PBX's and CO's just ring the PSTN line, they don't deliver the number. Make sure you Enable AutoMatic Dialing in the Endpoint Settings, and if you want the line in port x to be the number dialed to the sip side, datafill the number there. 3. Make sure you set up the audiocodes with the proper coder like Ulaw, they come set to 723 by default which is crap for coders. they can support up to 5 so just datafill them with all the big coders U, A, 729, and whatever else 4. The Advanced Configuration pages have all their Channel settings, make sure the fax's are set to what the Trixbox supports. Audiocodes by default does t.38 now. if your pbx isn't set up for it, you need to put the Audiocodes in a transparent or events mode If you want the source number from IP to use the same datafilled Endpoint Port on the PSTN side make sure Endpoint Phone Numbers has that number datafilled, and then set up a hunt group with source number as the selection algorithm(5.0). Assign the endpoints to that hunt group. IP to Tel rouitng route all calls to that group Endpoint Phone Number - This will give you the options for either 4 or 8 ports. You do not need to place anything here. However, it is a good idea to do such to help you identify which port the call comes in on; as you can view the reports in freePBX to identify calls. In my case, since I have four PSTN ports, I used the last four digits of the telephone number to identify. Identifying which PSTN line the call came from only works if you DO NOT have caller id on the line, or your turn off caller id. If caller ID is turn on, then freePBX will only record the receiving number.not the line number. Endpoint Settings - Automatic Dialing - Define a station number located on Asterisk / Trixbox (ie 101) for all ports - Caller ID - Allowed .. turn off if you want to Identify the line they came in on. - Detect Caller ID from Tel - Enable Thanks, Steve Totaro From: Angel Heart [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO Date: Mon, 26 Feb 2007 19:22:53 -0800 (PST) Hi José, I have not resolve this issue yet. I am currently focusing in my newly arrived toy (fonebridge2)
RE: [asterisk-users] To use asterisk or proprietary hardware, that is the questio
From: shadowym [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [asterisk-users] To use asterisk or proprietary hardware,that is the question Date: Mon, 26 Feb 2007 20:42:21 -0800 Thanks Tom and everyone else, Based largely on your comments I decided to just stick with what works. I have a site using entry level ATX server hardware that has been solid as a rock. I'll just go with that instead of more specialized fanless hardware, specialized power supply and 2.5 hard drives etc. Maybe get a second motherboard as a spare of they go for the ongoing remote support option. I'll do some simple things like a put in a standby hard drive with the production image on it in case the primary drive fails. The case has hot swap SATA bays so if the primary drive fails or get's corrupted anyone can just swap drives and they will be back up just like that. I'll make remote offsite backups as well. Thanks for all the help. -Original Message- From: Tom [mailto:[EMAIL PROTECTED] Sent: Saturday, February 24, 2007 5:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] To use asterisk or proprietary hardware,that is the question At 11:53 AM 2/24/2007, you wrote: Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a set it and forget it type of thing like a lot of small business proprietary systems. There is no such thing as set and forget. Businesses change. They either grow or shrink, they don't stand still. They will add and remove phones. So they will call you at that time. Or are you expecting them to shop for their own phones on Ebay? At the same time they would like to be able to do minor dial plan changes themselves so I have determine that a GUI like FreePBX or similar alternative (free or commercial) is appropriate. We take a different approach. We don't want a GUI. We don't want the limits. We work with the business to design their dial plan. Then we write it. We do not give them a GUI because we don't want them making changes and then asking for support. We sell them a minor service agreement and remote in for any changes. We also handle professional voice recording and basic training on phone use. And we handle backups and service if needed. Once they understand that we can do that without a service call, they are quite receptive to the idea. Conventional PBXs come with service agreements so customers are used to that but surprised at the low cost from you. I have some concerns about using Asterisk for this. As much as I am in support of the whole Asterisk revolution, I just do not feel confident enough in Asterisk on a Hard Drive as a set it and forget it setup running month after month, year after year. I am hoping someone can convince me otherwise. Hard drives are reliable. But I have similar feelings so we are working on a flash solution. Were running it beta in our office right now. It only uses the hard drive for daily voicemail, boots from flash and runs from RAM. I'm concerned about hard drive corruptions/failures, memory leaks, software bugs etc. Conventional systems have bugs too. I have the budget to buy good quality hardware so if I was to go with Asterisk I would go industrial grade fanless computer, power conditioned UPS etc. You don't really need fanless. Make it cheap enough that it can easily be replaced. Like a $500 PC. I am not concerned about the reliability of most of the hardware. It's the hard drive and the software that runs on it that worries me. I will obviously use a mature stable Asterisk release and the most stable Linux version which I won't bother naming just to keep the discussion focussed. Asterisk is pretty darn stable. I have other Asterisk installs that went well but they were in environments where there were IT people around who were prepared to deal with some Linux administration and I could provide ongoing support for more major things. That is not the case here. Some of those sites have been running for months untouched, some needed some updates and reboots for various issues. I don't think this customer would look very favorably on me having to come in and add patches or have to reboot once a month or whatever. So do it from home. And how often do you really need to upgrade a minimal read only flash based system with no dev tools running from RAM? Does the latest kernel really matter? Their expection is the same as they would have with any other phone system that mounts on the wall and just works for years. I think that is a reasonable expectation. Agreed. And if it breaks, you replace it quickly and at a low cost. I am looking at
RE: [asterisk-users] To use asterisk or proprietary hardware, that is the questio
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Totaro Sent: Monday, February 26, 2007 11:11 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] To use asterisk or proprietary hardware,that is the questio From: shadowym [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [asterisk-users] To use asterisk or proprietary hardware,that is the question Date: Mon, 26 Feb 2007 20:42:21 -0800 Thanks Tom and everyone else, Based largely on your comments I decided to just stick with what works. I have a site using entry level ATX server hardware that has been solid as a rock. I'll just go with that instead of more specialized fanless hardware, specialized power supply and 2.5 hard drives etc. Maybe get a second motherboard as a spare of they go for the ongoing remote support option. I'll do some simple things like a put in a standby hard drive with the production image on it in case the primary drive fails. The case has hot swap SATA bays so if the primary drive fails or get's corrupted anyone can just swap drives and they will be back up just like that. I'll make remote offsite backups as well. Thanks for all the help. -Original Message- From: Tom [mailto:[EMAIL PROTECTED] Sent: Saturday, February 24, 2007 5:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] To use asterisk or proprietary hardware,that is the question At 11:53 AM 2/24/2007, you wrote: Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a set it and forget it type of thing like a lot of small business proprietary systems. There is no such thing as set and forget. Businesses change. They either grow or shrink, they don't stand still. They will add and remove phones. So they will call you at that time. Or are you expecting them to shop for their own phones on Ebay? At the same time they would like to be able to do minor dial plan changes themselves so I have determine that a GUI like FreePBX or similar alternative (free or commercial) is appropriate. We take a different approach. We don't want a GUI. We don't want the limits. We work with the business to design their dial plan. Then we write it. We do not give them a GUI because we don't want them making changes and then asking for support. We sell them a minor service agreement and remote in for any changes. We also handle professional voice recording and basic training on phone use. And we handle backups and service if needed. Once they understand that we can do that without a service call, they are quite receptive to the idea. Conventional PBXs come with service agreements so customers are used to that but surprised at the low cost from you. I have some concerns about using Asterisk for this. As much as I am in support of the whole Asterisk revolution, I just do not feel confident enough in Asterisk on a Hard Drive as a set it and forget it setup running month after month, year after year. I am hoping someone can convince me otherwise. Hard drives are reliable. But I have similar feelings so we are working on a flash solution. Were running it beta in our office right now. It only uses the hard drive for daily voicemail, boots from flash and runs from RAM. I'm concerned about hard drive corruptions/failures, memory leaks, software bugs etc. Conventional systems have bugs too. I have the budget to buy good quality hardware so if I was to go with Asterisk I would go industrial grade fanless computer, power conditioned UPS etc. You don't really need fanless. Make it cheap enough that it can easily be replaced. Like a $500 PC. I am not concerned about the reliability of most of the hardware. It's the hard drive and the software that runs on it that worries me. I will obviously use a mature stable Asterisk release and the most stable Linux version which I won't bother naming just to keep the discussion focussed. Asterisk is pretty darn stable. I have other Asterisk installs that went well but they were in environments where there were IT people around who were prepared to deal with some Linux administration and I could provide ongoing support for more major things. That is not the case here. Some of those sites have been running for months untouched, some needed some updates and reboots for various issues. I don't think this customer would look very favorably on me having to come in and add patches or have to reboot once a month or whatever. So do it from home. And how often do you really need to upgrade a minimal read only flash based system with no dev tools running from RAM? Does
Re: [asterisk-users] How to get values of local channels context
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Context macro-test) in
new stack
I want to get 116-2000 somehow.
Any suggestions would be appreciated.
So use ${MACRO_CONTEXT} .
Thanks
But doesn't this give the calling context which, if itself is another macro,
will still not give me what I want? If macro-test is called by macro-first then
${MACRO_CONTEXT} = macro-first. Surely there's a way to get the context
directly from the Local channel itself?
Cameron
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Re: [asterisk-users] How to get values of local channels context
From: kjcsb [EMAIL PROTECTED]
Date: Mon, 26 Feb 2007 22:32:29 -0800 (PST)
CLI shows:
-- Executing NoOp(Local/[EMAIL PROTECTED],2, Context
macro-test) in new stack
I want to get 116-2000 somehow.
Any suggestions would be appreciated.
So use ${MACRO_CONTEXT} .
Thanks
But doesn't this give the calling context which, if itself is another
macro, will still not give me what I want? If macro-test is called by
macro-first then ${MACRO_CONTEXT} = macro-first. Surely there's a way to
get the context directly from the Local channel itself?
Cameron
If nested macro calls are necessary, define an inheritable local variable,
e.g., __real-context. Two _'s enables infinite inheritance. Hope this
helps.
Yuan Liu
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[asterisk-users] chan_sip.c:10173 handle_response: Dont know how to handle a 202 Accepted respons
What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) Yuan Liu ___ --Bandwidth 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 - Streaming Audio Bridge
I used mpg123 to stream air traffic control as a MOH class but I also found it didn't always work with the shoutcast servers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Germann Sent: 27 February 2007 02:17 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Asterisk - Streaming Audio Bridge Greetings, Does anyone know of a tool that can act as a VoIP client and stream to a streaming server such as shoutcast/icecast, etc. I've got a client interested in doing basketball play by plays during tourney season. They have * in place now and the bandwidth to burn for streaming out. In the old world, I did an analog phone patch - mixer - encoder - streaming server. What I'm thinking of is more along the lines of a client that registers as a SIP/IAX client, answers the phone and patches it to a streaming server. Thoughts/suggestions? Thanks Eric ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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
