[asterisk-users] chan_oh323 early media
Hi, I configured openh323_v1_18_0, pwlib_v1_10_0 and asterisk-oh323-0.7.3. I can call inbound and outbound. But early media is not working in outboubd. Regards, Jason. oh323.conf == [general] listenPort=1720 connectPort=1720 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 ;fastStart=yes fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=yes jitterMin=20 jitterMax=500 ipTos=reliability outboundMax=20 inboundMax=20 ;bandwidthLimit=1024 wrapLibTraceLevel=10 libTraceLevel=10 ;wrapLibTraceLevel=0 ;libTraceLevel=0 libTraceFile=stdout gatekeeper=192.168.1.150 gatekeeperTTL=60 ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 userInputMode=TONE amaFlags=billing accountCode=aaabbbaaabbb context=from-323 [register] context=from-323 alias=MyH323ID alias=555 alias=5556667 alias=5556668 [codecs] codec=G711A frames=20 codec=G711U frames=20 codec=G7231 frames=20 codec=G729A frames=20 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and mysql
Dear All, I' I have a problem in installing asterisk 1.4.0. how can i compile res_config_mysql.c in astersisk-addons dir. I've downloaded asterisk-addons-1.4.0 compiling and installing it. But i can't found shared object of res_config_mysql.so. My system is : Debian Linux 3.1 Kernel 2.6.8-11 asterisk-1.4.0 zaptel-1.4.0 asterisk-addons-1.4.0 libmysqlclient using apt-get webserver : xampp with mysql builtin and included in xampp error log is : [Jan 2 16:51:56] WARNING[30714] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: cannot open shared object file: No such file or directory. what is the missing step ? can anyone help me ? ___ --Bandwidth 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] Realtime multiple registration for a Hard Phone Snom 360
2006/12/29, Frédéric Marti [EMAIL PROTECTED]: Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can someone help for this problem ? Regards ** Fred * * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Fred, Are you sure Asterisk handles multiline registrations ? Could it be a Snom feature needing another call manager to happen ? Regards ___ --Bandwidth 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 mysql
slave, have you install asterisk-addons yet? if you installed and that error still happen, pls find that file, you can also put them into that path to load. regards, osochebol - Original Message From: RdBSD [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 2, 2007 1:57:15 AM Subject: [asterisk-users] asterisk and mysql Dear All, I' I have a problem in installing asterisk 1.4.0. how can i compile res_config_mysql.c in astersisk-addons dir. I've downloaded asterisk-addons-1.4.0 compiling and installing it. But i can't found shared object of res_config_mysql.so. My system is : Debian Linux 3.1 Kernel 2.6.8-11 asterisk-1.4.0 zaptel-1.4.0 asterisk-addons-1.4.0 libmysqlclient using apt-get webserver : xampp with mysql builtin and included in xampp error log is : [Jan 2 16:51:56] WARNING[30714] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: cannot open shared object file: No such file or directory. what is the missing step ? can anyone help me ? ___ --Bandwidth 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] [OT] Wifi SIP phones - LinkSys WIP330
Maybe, what is meant is handover. Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Buying
Dear Guys Merry Christmas and happy new year . Please do any one knows from where I can buy a full pbx corporate cd and integrated with exchange server and life communication server . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth 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 + door opener
Dovid B wrote: can u get me the info on the part ? Hi Guys I have found this. Have not tested as yet, but have asked them for some more info. Might be of some help. www.its-tel.com Cheers Terry ___ --Bandwidth 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] Problem with centos 4.4 and jabber/gtalk (really iksemel)
Kenneth Padgett wrote: I'm working from the docs here: http://voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk and getting an error doing the ./configure on the iksemel module: checking for getaddrinfo... yes ./configure: line 20399: syntax error near unexpected token `,' ./configure: line 20399: `AM_PATH_LIBGNUTLS(,' It seems to want the libgnutls-dev package as per the documentation. Problem is, I can't seem to find such a package for centos 4.4. Anyone have any advice? Thanks! -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Kenneth, It looks like the gnutls development package is called gnutls-devel: Available Packages Name : gnutls-devel Arch : x86_64 Version: 1.0.20 Release: 3.2.3 Size : 503 k Repo : update Summary: Development files for the gnutls package. Description: The GNU TLS library implements TLS. This package contains files needed for developing applications with the GNU TLS library. Someone needs to fix this description. 'yum install gnutls-devel' should get the package installed. Bob... ___ --Bandwidth 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] Avoiding deadlock-line drop problem
Hi,all Randomly my line drops and when I look in message log file I always see the following notice: NOTICE[14491] chan_zap.c : avoiding deadlock… The situation appears with no obvious reason, the CLI shows nothing more than the zaptel channel hanging up. I have a Asterisk 1.2.10 and Zaptel 1.2.7 installation on a MSI motherboard with intel chipset 915G.The machine is equiped with a TDM40B and a TDM22B. Can somebody help with this mess? ___ --Bandwidth 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] Best Hardware for Asterisk Server?
Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? With desktop prices so cheap you might think that you should just buy them off the shelf, but is that really a reliable machine? Anything you can tell me that would assist me in deciding the best way to obtain and maintain these boxes would be very helpful. I have even looked into building system myself that have no moving parts, but for about the same price I can build an immensely more powerful machine WITH moving parts. - Mark ___ --Bandwidth 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] Save SIP DEBUG output to a file
Hi guys, How can i save sip debug command output to a file ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI ANI/CallerID
add a wait before you dial the sip phone, keep in mind the callerid information arrives later than the call setup info On Dec 31, 2006, at 4:38 PM, David Sampson wrote: For some reason something that seems like it should be simple is leaving me a bit perplexed. I am receiving incoming CallerID ANI on my PRI, but on my VoIP phones the display just shows asterisk when calls come in. I am receiving the calls with DNIS and have the DNIS digits setup as extensions. Do I need to add something to force relay the received caller ID to the phone? Any help is appreciated... Thanks, Dave ___ --Bandwidth 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] [OT] Wifi SIP phones - LinkSys WIP330
Noah is correct. We will install a trial system with 11 AP. The WiFi terminal will hold a conversation when moving between APs. Initial tests with Hitachi IP5000 were ok. We need to test as well PDA and cell/WiFi phones. Jorge Mendoza Noah Miller wrote: Roaming is irrelevant in VOIP. You just need a fairly good wifi connection. I don't think they mean roaming in traditional cell-phone terms. I think they mean moving between different Access Points on a single WiFi network. Judging by the reports in this thread, some Wifi phones do this better than others. ___ --Bandwidth 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 mysql
I had this same problem. It was that I was missing the mysql-devel package. I installed this on my Fedora Core 4 system with yum install mysql-devel. Once I installed this I redid the ./configure, make and make install of the addons and voila it was there. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ngo Duc Loi Sent: Tuesday, January 02, 2007 4:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and mysql slave, have you install asterisk-addons yet? if you installed and that error still happen, pls find that file, you can also put them into that path to load. regards, osochebol - Original Message From: RdBSD [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 2, 2007 1:57:15 AM Subject: [asterisk-users] asterisk and mysql Dear All, I' I have a problem in installing asterisk 1.4.0. how can i compile res_config_mysql.c in astersisk-addons dir. I've downloaded asterisk-addons-1.4.0 compiling and installing it. But i can't found shared object of res_config_mysql.so. My system is : Debian Linux 3.1 Kernel 2.6.8-11 asterisk-1.4.0 zaptel-1.4.0 asterisk-addons-1.4.0 libmysqlclient using apt-get webserver : xampp with mysql builtin and included in xampp error log is : [Jan 2 16:51:56] WARNING[30714] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: cannot open shared object file: No such file or directory. what is the missing step ? can anyone help me ? ___ --Bandwidth 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] Slightly updated UK English voice prompts
I believe there were some new prompts added for 1.4 for Directory Info. These have now been added to http://www.tel.net Have a good 2007. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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] (OT) Where to post free source for AGI?
Happy Holidays! Sourceforge provides free hosting for open source projects. That is where I would put it if I were me. For licensing.. I use the BSD license for my creations, but version 2 of the GPL is stronger in my opinion. Good luck, Ejay Hire -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Sunday, December 31, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] (OT) Where to post free source for AGI? Hey all, After figuring out a problem with AGI and freepascal, I have finished writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt. I'd like to give it to the community (source/binary) and was wondering where to post it? The wiki? Also, anyone have suggestion on licensing? LGPL? FreeBSD? Thanks -- Warm Regards, Lee ___ --Bandwidth 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] 802.1x support in wired sip hardphones ?
Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards ___ --Bandwidth 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] (OT) Where to post free source for AGI?
After skimming over your readme file I thought I would ask, how does this app differ from passing the parameters to the swift program using a System dial plan command? You mention having cepstral play back a text file in a certain voice, which I have done from the dialplan with the options provided by cepstral. I just want to see if I missed something. On 12/31/06, Lee Jenkins [EMAIL PROTECTED] wrote: Hey all, After figuring out a problem with AGI and freepascal, I have finished writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt. I'd like to give it to the community (source/binary) and was wondering where to post it? The wiki? Also, anyone have suggestion on licensing? LGPL? FreeBSD? Thanks -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth 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] (OT) Where to post free source for AGI?
I would post it to some site of yours (or Sourceforge if you plan to have shared CVS) plus a page on the wiki, so people can find it. I have been working on a few projects on sourceforge and never had problems with it. With licence, you choose. GPL is usually a good starting point for licensing. l. On Sun, 31 Dec 2006 18:44:48 +0100, Lee Jenkins [EMAIL PROTECTED] wrote: Hey all, After figuring out a problem with AGI and freepascal, I have finished writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt. I'd like to give it to the community (source/binary) and was wondering where to post it? The wiki? Also, anyone have suggestion on licensing? LGPL? FreeBSD? Thanks -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.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] 802.1x support in wired sip hardphones ?
Hi, http://www.communications.siemens.co.uk/enterprise/products/optiPoint_410s.htm rich. --- Olivier [EMAIL PROTECTED] wrote: Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) Where to post free source for AGI?
On Sun, Dec 31, 2006 at 12:44:48PM -0500, Lee Jenkins wrote: Hey all, After figuring out a problem with AGI and freepascal, I have finished writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt. I'd like to give it to the community (source/binary) and was wondering where to post it? The wiki? Please put a pointer to whereever the code is from the wiki, yes. Also, anyone have suggestion on licensing? LGPL? FreeBSD? FreeBSD is a software distribution, not a software license. I guess you refer to the (modified, a.k.a 3-clause) BSD license. A similar and simpler license is the MIT (original X11) license. I personally prefer the GPL, but that is a matter of preference, and this is your code to license. Just post it and link to it. If there is some public interest, consider using a facility like SourceForge to enable others to easily help with the development. -- 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
Re: [asterisk-users] 802.1x support in wired sip hardphones ?
Thanks !! I've never heard of this one (I mean : I've never heard of OptiPoint phones to support 802.1x). Have you used the SIP version with Asterisk and 802.1x ? Am I correct to think that using 802.1x isn't directly of Asterisk concern ? 2007/1/2, richard Coco [EMAIL PROTECTED]: *** This message was sent to your KasMail disposable email address: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *** Hi, http://www.communications.siemens.co.uk/enterprise/products/optiPoint_410s.htm rich. --- Olivier [EMAIL PROTECTED] wrote: Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Hardware for Asterisk Server?
--- Mark Greene [EMAIL PROTECTED] wrote: Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? (In the US) I have had very good luck with Opterons in Tyson rackmounts bought from Newegg. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 802.1x support in wired sip hardphones ?
--- Olivier [EMAIL PROTECTED] wrote: Thanks !! I've never heard of this one (I mean : I've never heard of OptiPoint phones to support 802.1x). Have you used the SIP version with Asterisk and 802.1x ? we have several Optipoint410/420/600 configured with Asterisk and they seem to work well (but no 802.1x).We made several tests with MacAuthentication last year. Am I correct to think that using 802.1x isn't directly of Asterisk concern ? 802.1x has nothing to do with Asterisk. You need a supplicant (your phone) an Authenticator (your switch) and a authentication server (e.g FreeRadius) a howto about 802.1X Port-Based Authentication are avalaible at http://tldp.org/HOWTO/html_single/8021X-HOWTO/ 2007/1/2, richard Coco [EMAIL PROTECTED]: *** This message was sent to your KasMail disposable email address: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *** Hi, http://www.communications.siemens.co.uk/enterprise/products/optiPoint_410s.htm rich. --- Olivier [EMAIL PROTECTED] wrote: Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Hardware for Asterisk Server?
On Tue, 2 Jan 2007, Mark Greene wrote: Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? With desktop prices so cheap you might think that you should just buy them off the shelf, but is that really a reliable machine? Anything you can tell me that would assist me in deciding the best way to obtain and maintain these boxes would be very helpful. I have even looked into building system myself that have no moving parts, but for about the same price I can build an immensely more powerful machine WITH moving parts. The best hardware is the hardware that you're most familiar with - the hardware they you know will be reliable and know how to fix it if/when it goes wrong. And yes, you do end up paying slightly more (sometimes) for smaller, quieter, and no-moving parts kit. It's all to do with volume of sales I guess! If you have a computer/comms room with servers, etc. already in-place, then noise isn't going to be an issue for you, but you still want reliability. So if you are having moving parts (ie. disks!) then get two and run them in a RAID-1 (mirror) configuration. Think about redundant PSUs. (and UPS - and UPS the Ethernet switch, and think about PoE) Fit good ball bearing fans and if building it yourself, good thermal grease. Soaktest the system before it goes live. For some production machines, I'm using mini-ITX boards - 1GHz processors, fanless, diskless (boot off flash) but they aren't without their limitations (I doubt they'd be happy in a 100-extension office for example ;-) But I am currently looking at a 150-line system, but I'm still going to boot it off flash, just to reduce one failure point in the system... 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
[asterisk-users] How to show a debugging remark in a sip or extensions context?
I would like to show a remark that would show call progress and appear on the CLI screen. The remark should be in the code of a sip [channel] or extentions [context] If I can't send my own remark, what little used 'show' command could I insert in the code? Can this be done? -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth 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] (OT) Where to post free source for AGI?
Also, anyone have suggestion on licensing? LGPL? FreeBSD? One advantage of LGPL over GPL is that GPL is 'viral' whereas LGPL is not. For a more in depth discussion please see: http://www.ugcs.caltech.edu/manuals/devtool/autotoolset-0.11.4/toolsmanual_87.html In short, if you want anyone to be able to distribute your software within their own packages, even proprietary and/or commercial ones, then use the LGPL. If you want your software to follow the tenets of 'free and open source' more strictly, then use the GPL. Both licenses protect your software, but they place different limits on how the software is distributed. Hope this helps! -MC ___ --Bandwidth 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 connect two asterisk server
Your best bet is to contact Sysmaster support at [EMAIL PROTECTED] or 1877-900-3993. I was talking to one of our contacts there and he said that it would be best to have you contact them. In order to get it to work for you they need to know the exact configuration you are trying to set up. We've worked with Sysmaster for some time now and they are very nice and helpful people. -David Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com On Jan 1, 2007, at 8:53 PM, Noah Miller wrote: Hi Again Dan - Its a VOIP Switch based on SIP Proxy. Its Voice Master from SysMaster. VoiceMaster only authenticates IP and cant have username password based authentication which asterisk can do. So i need to take some traffic from VoiceMaster to Asterisk and terminate it. That shouldn't be a problem. You can just create a sip friend/peer without a username or password, and with a host=ipaddress statement. Like this in your sip.conf file: [NoAuth-VoiceMaster] type=friend context=your context host=IP Address Of Voice Master disallow=all allow=codecs you want to allow As an addendum to this, it would be a very good idea to make certain that you've properly secured your asterisk server so you're not going to have unwanted unauthorised access. I would probably only do this if your asterisk server is not accessible from the outside world via sip. - Noah ___ --Bandwidth 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] Best Hardware for Asterisk Server?
I believe I am going to start out with some refurbished Dell Poweredge servers. They have had a high success rate with a friend. ___ --Bandwidth 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] chan_oh323 early media
I am a a little confused on how to get h323 working on asterisk. Could you please point me towards specific resources you used? voip-info.org seems to keep me in a loop of info. ___ --Bandwidth 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] Best Hardware for Asterisk Server?
Mark Greene[EMAIL PROTECTED] Wrote on: 1/2/2007 12:58 PM: I believe I am going to start out with some refurbished Dell Poweredge servers. They have had a high success rate with a friend. I was going to go that route as well. But, depends on the model. I have several of the Poweredge 2300/2400 variety and these seem problematic. I could not get the final compile steps to perform on the 2400, for instance. Forget the exact issue. Also, these models, at least, do not directly support IDE drives, such as CD/DVD items. You are limited to SCSI versions, or trying to hack in an IDE controller. Which is fine, I guess, if all your source/install software is on CD. Or until the CDdrive fails and you have to hunt up a SCSI version. I've not seen, at any price, scsi versions of DVD drives. I am looking at the ACARD AEC7720-U IDE-SCSI bridge (converter) to get over that. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) Where to post free source for AGI?
Bruce Reeves wrote: After skimming over your readme file I thought I would ask, how does this app differ from passing the parameters to the swift program using a System dial plan command? You mention having cepstral play back a text file in a certain voice, which I have done from the dialplan with the options provided by cepstral. I just want to see if I missed something. The end result (ie: playing a voice) is the same, but the implementation is different. For me at least, that is significant in two ways: 1. Simply, easier to read syntax from the dialplan so it's easier for me to maintain. 2. Because there is a middle layer involved (the AGI), I can also group related system commands with in a single call. That means I can do more in a single dialplan line and try to maintain a more modular system. 3. By standardizing a call from the dialplan using an AGI script, I also help to insulate myself from changes that might happen to the cepstral API later on. I would just have to distribute a new agi to my installations and not have to change every reference to it in my their dial plans which is often error prone endeavor. I could also support different speech engines by swapping out the AGI app and never have to change my dial plan, assuming the parameters could be generic enough..? But, in the end, you're right, you can do the same thing from the system application. -- Warm Regards, Lee ___ --Bandwidth 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] SpanDSP and Asterisk 1.4
Has anyone made this combination work together? I've tried everything and can't seem to get it work right. It all compiles fine, but when rxfax is called, I get an unknown symbol error. From my reading, everything points to me having multiple copies of spandsp and it's maybe calling the wrong one. I went through the directories and they all look clean when I install. Here's what I'm trying: Asterisk-1.4.0 spandsp-20061217 (from the snapshots) The patchfile from the snapshots works except for one hunk, so I manually apply that one part. Anyone got this working? Any pointers? I had a previous copy of spandsp-0.0.2pre26 prior to this but I really think I got it all removed. Thanks! Mark ___ --Bandwidth 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] Best Hardware for Asterisk Server?
I believe I am going to start out with some refurbished Dell Poweredge servers. They have had a high success rate with a friend. One word of caution: some have had various hardware issues getting certain telephony cards to work with certain Dell PowerEdge servers. If you aren't going to have telephony cards in your system, i.e. VoIP-only setup, then you're probably good to go. If not, do a list search on Dell PowerEdge and review the feedback given by those who've already been where you are now. Hopefully their experience will save you time, money, and the occasional headache! -MC ___ --Bandwidth 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 connect two asterisk server
Hi all, Special thanks to David and Noah for the earnest efforts... Dan On 02/01/07, Dave Schardin [EMAIL PROTECTED] wrote: Your best bet is to contact Sysmaster support at [EMAIL PROTECTED] or 1877-900-3993. I was talking to one of our contacts there and he said that it would be best to have you contact them. In order to get it to work for you they need to know the exact configuration you are trying to set up. We've worked with Sysmaster for some time now and they are very nice and helpful people. -David Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com On Jan 1, 2007, at 8:53 PM, Noah Miller wrote: Hi Again Dan - Its a VOIP Switch based on SIP Proxy. Its Voice Master from SysMaster. VoiceMaster only authenticates IP and cant have username password based authentication which asterisk can do. So i need to take some traffic from VoiceMaster to Asterisk and terminate it. That shouldn't be a problem. You can just create a sip friend/peer without a username or password, and with a host=ipaddress statement. Like this in your sip.conf file: [NoAuth-VoiceMaster] type=friend context=your context host=IP Address Of Voice Master disallow=all allow=codecs you want to allow As an addendum to this, it would be a very good idea to make certain that you've properly secured your asterisk server so you're not going to have unwanted unauthorised access. I would probably only do this if your asterisk server is not accessible from the outside world via sip. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] yet another faxing issue (outbound only, via ATA)
2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great! Never had an issue The problem, however is outbound. Sipura 1001 ATAs. Fax machine connected to the ATA. Registered to the 2nd asterisk box. Keep in mind this server runs voice calls just fine. Outbound calls from this box are ulaw The call is then sent via IAX2 (also tried SIP as well) to the Asterisk server w/ the PRI, then out to the world Hit and miss to send faxes out Echo cancellation is enabled on the PRI I have lowered the rxgain and txgain to -5.0, seems fine for voice. The ATA is running 3.1.8 firmware from Sipura with fax detect turned Usually the faxes fail, but sometimes you will get all the pages, but only a fraction of the page. I have tried turning off ECM but still the same issue. I would suspect the Sangom or IAX2, or something of that nature except receiving faxes traveling to the 2nd asterisk box works just fine! I also tried to register the ATA to the primary Asterisk server w/ the PRI, same exact issue. Any ideas - better luck w/ Grandstream? I suspect the problem is not Asterisk, or the Sangoma, or jitter or bandwidth since receiving faxes works fine. I did not try to receive faxes through the ATA to the machine itself, I tried that a few months ago during other testing, never got it to work so I never tried again once I got NVFaxDetect working for email. My next step is to connect the fax machine to a Wildcard X100P. Any other suggestions? Black magic? Voodoo? 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
[asterisk-users] Re: Hi reg. 2 asterisk server
Hi Thiru -
Please clarify one more doubt in extensions.conf file...
is the following dial plan is right way to call another server(frome
serverA to serverB)
exten = _5X,1,Dial(sip/[EMAIL PROTECTED]:6030,15,tr)
exten = _5X,2,Hangup
You can dial either via IP or by sip device name (the name in brackets
[] in sip.conf). Either way, you also have to include the username
and password in the Dial statement(). It looks like this:
exten = _5X,1,Dial(SIP/UserB:[EMAIL PROTECTED]/${EXTEN},15,tr)
or
exten = _5X,1,Dial(SIP/UserB:[EMAIL PROTECTED]/${EXTEN},15,tr)
If you want to avoid putting the password in the dialplan, and make
the authentication process a little more secure, you can also use MD5
authentication. This page on the wiki explains how:
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+md5secret
- Noah
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Re: [asterisk-users] Best Hardware for Asterisk Server?
At 07:20 1/2/2007, Mark Greene, wrote: Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? With desktop prices so cheap you might think that you should just buy them off the shelf, but is that really a reliable machine? Anything you can tell me that would assist me in deciding the best way to obtain and maintain these boxes would be very helpful. I have even looked into building system myself that have no moving parts, but for about the same price I can build an immensely more powerful machine WITH moving parts. - Mark Case: 1 CodeGen 4U Server Case $80 http://tinyurl.com/bnobz http://tinyurl.com/95s2b Power Supply: 1 Dual 450 W. Power supply -- IStar https://www.ewiz.com/detail.php?name=PS-TC50R8A http://www.directron.com/tc400r8.html Motherboard, CPU 2GB of memory: http://www.mwave.com/mwave/skusearch.hmx?scriteria=MB-BA23083 http://www.mwave.com/mwave/viewspec.hmx?scriteria=BA21409 2 Hard Drives in RAID 1 config: http://www.mwave.com/mwave/viewspec.hmx?scriteria=AA48770 Digium card: 2 port, 64 bit, 3.3 volt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Grandstream GXW-4108 8 port FXO
On 2006-12-21 13:29:47 -0800, cb [EMAIL PROTECTED] said: Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using similar devices (that is, FXO devices that connect to the Asterisk server via SIP over Ethernet). I am looking to connect at least 8 PSTN lines, and as many as 12 or 16 to Asterisk (Currently using Trixbox, but I'm also looking at either AsterixNow or just building from scratch on a bare linux box). Money is a major concern in my purchases, which is why I'm looking at the Grandstream (even used on ebay, I don't seem to be able to find 8-16 port FXO devices for less than the approx $50 per port the Grandstream will get me... plus it has a video input for a security camera which is just a plus to me as installing a web capable surveillance camera at the location is on my to do list). You get what you pay for. I originally deployed the GS HT-488 as an FXO gateway, as I also thought that was a good deal. Turns out, I would have been much better off paying four times as much (which I did end up doing), to get something that actually works reliably. Sometimes a good deal isn't. Then again, I have no experience with the particular piece you mention. Personally I would never spend another penny on any grandstream product after dealing with the HT-488s FXO. Marty ___ --Bandwidth 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 connected, cannot hear or speak - $20 for fix
I am able to get this script to dial, but I am unable to talk or hear
anything. The script asks for the number to call and the the caller id to
display (if user is not at their normal extension). Once submitted, the
external extension receives a call, once answered the call is then placed to
the dentition number.
The script works as the call is place, but I cannot hear or say anything.
Any one that is able to get this going I would be will to give $20 to (via
paypal)
html
head
titlecall/title
/head
body
?
#--
#edit the below variable values to reflect your system/information
#--
#specify the name/ip address of your asterisk box
#if your are hosting this page on your asterisk box, then you can use
#127.0.0.1 as the host IP. Otherwise, you will need to edit the following
#line in manager.conf, under the Admin user section:
#permit=127.0.0.1/255.255.255.0
#change to:
#permit=127.0.0.1/255.255.255.0,xxx.xxx.xxx.xxx ;(the ip address of the
server this page is running on)
$strHost = 127.0.0.1;
#specify the username you want to login with (these users are defined in
/etc/asterisk/manager.conf)
#this user is the default AAH AMP user; you shouldn't need to change, if
you're using AAH.
$strUser = admin;
#specify the password for the above user
$strSecret = amp111;
#specify the channel (extension) you want to receive the call requests with
#e.g. SIP/XXX, IAX2/, ZAP/, etc
$strChannel = Local/[EMAIL PROTECTED];
#specify the context to make the outgoing call from. By default, AAH uses
from-internal
#Using from-internal will make you outgoing dialing rules apply
$strContext = from-internal;
#specify the amount of time you want to try calling the specified channel
before hangin up
$strWaitTime = 30;
#specify the priority you wish to place on making this call
$strPriority = 1;
#specify the maximum amount of retries
$strMaxRetry = 2;
#
#Shouldn't need to edit anything below this point to make this script work
#
#get the phone number from the posted form
$strExten = $_POST['txtphonenumber'];
#specify the caller id for the call
$strCallerId = $_POST['txtcid'];
$length = strlen($strExten);
if ($length == 11 is_numeric($strExten))
{
$oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection
to host failed);
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Events: off\r\n);
fputs($oSocket, Username: $strUser\r\n);
fputs($oSocket, Secret: $strSecret\r\n\r\n);
fputs($oSocket, Action: originate\r\n);
fputs($oSocket, Channel: $strChannel\r\n);
fputs($oSocket, WaitTime: $strWaitTime\r\n);
fputs($oSocket, CallerId: $strCallerId\r\n);
fputs($oSocket, Exten: $strExten\r\n);
fputs($oSocket, Context: $strContext\r\n);
fputs($oSocket, Priority: $strPriority\r\n\r\n);
fputs($oSocket, Action: Logoff\r\n\r\n);
fclose($oSocket);
?
p
table width=300 border=1 bordercolor=#63 cellpadding=3
cellspacing=0
trtd
font size=2 face=verdana,georgia color=#63We are currently
trying to call you. Please be patient, and wait for
your phone to ring!brIf your phone does not ring after 2 minutes, we
apologize, but must either be out, or
already on the
phone.bra href=? echo $_SERVER['PHP_SELF'] ?Try Again/a/font
/td/tr
/table
/p
?
}
else
{
?
div align=center
table width=300 border=1 bordercolor=#63 cellpadding=3
cellspacing=0
tr
tdform action=? echo $_SERVER['PHP_SELF'] ? method=post
p align=leftfont size=2 face=verdana,arial,georgia
color=#63Enter number to call (11 Digits):/font/p
p align=center
input type=text size=20 maxlength=11
name=txtphonenumber
/p
pfont size=2 face=verdana,arial,georgia
color=#63Enter your caller ID/font:/p
p align=center
input type=text size=20 maxlength=11 name=txtcid
/p
p
input type=submit value=Make Call
/p
/form /td/tr
/table
/p
?
}
?
/div
/body
/html
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Re: [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really iksemel)
Bob, It looks like the gnutls development package is called gnutls-devel: 'yum install gnutls-devel' should get the package installed. Yah, I thought that would be it. I have that installed, as well as gnutls. (I basically installed both packages you can find with yum search gnutls). Any other thoughts, can I just d/l the libs and uncompress them somewhere? -Kenneth ___ --Bandwidth 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] Best Hardware for Asterisk Server?
ASUS motherboards, in particular, have worked for me perfectly, everytime with both Digium and Sangoma cards. They are also easy to work with and well documented. -Original Message- From: Doug [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 02, 2007 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Hardware for Asterisk Server? At 07:20 1/2/2007, Mark Greene, wrote: Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? With desktop prices so cheap you might think that you should just buy them off the shelf, but is that really a reliable machine? Anything you can tell me that would assist me in deciding the best way to obtain and maintain these boxes would be very helpful. I have even looked into building system myself that have no moving parts, but for about the same price I can build an immensely more powerful machine WITH moving parts. - Mark Case: 1 CodeGen 4U Server Case $80 http://tinyurl.com/bnobz http://tinyurl.com/95s2b Power Supply: 1 Dual 450 W. Power supply -- IStar https://www.ewiz.com/detail.php?name=PS-TC50R8A http://www.directron.com/tc400r8.html Motherboard, CPU 2GB of memory: http://www.mwave.com/mwave/skusearch.hmx?scriteria=MB-BA23083 http://www.mwave.com/mwave/viewspec.hmx?scriteria=BA21409 2 Hard Drives in RAID 1 config: http://www.mwave.com/mwave/viewspec.hmx?scriteria=AA48770 Digium card: 2 port, 64 bit, 3.3 volt ___ --Bandwidth 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] Best Hardware for Asterisk Server?
Wow Doug thanks for the specs. This has really helped. ___ --Bandwidth 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] (OT) Where to post free source for AGI?
Thanks to all for the feedback. I have created a wiki page here: http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper http://preview.tinyurl.com/yl9utq I will host it on my company website for now. Seems like a small project to bother with SF.net. -- Warm Regards, Lee ___ --Bandwidth 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] (OT) Where to post free source for AGI?
- Lee Jenkins [EMAIL PROTECTED] wrote: Thanks to all for the feedback. I have created a wiki page here: http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper http://preview.tinyurl.com/yl9utq I will host it on my company website for now. Seems like a small project to bother with SF.net. -- Warm Regards, Lee Why not just post the text of the AGI to the wiki page? -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: yet another faxing issue (outbound only, via ATA)
Follow up: I used my Cisco 3660 that's a hop away and connected to a different PRI provider. Faxes work _fine_ From the ATA box I faxed a DID that would come back into the Zap enabled Asterisk server, then talks BACK to the server that the ATA box is regstered via IAX2 (or SIP, I found they both worked) and was able to receive the fax fine (incoming fax went to email) FAILURE FAX: Here is the path: ATA - SIP - * - IAX2 (or SIP) - * with Zap - send call out via Zap channel SUCCESS FAX: ATA - SIP - * - IAX2 (or SIP) - * with Zap - SIP to 3660 then out via PRI works every time! I tried G3 and ECM mode. ECM was flakey work even through the 3660 but G3 worked everytime. I have set the fax machine to G3 for the time being since it works each time. Each outbound call actually initiates a call to a DID that terminates into my * with the Zap card, then talks via IAX2 again back to the original server. No problems there. So I know that faxing other fax machines fails so it's not necessarily that there is some weird loop calling out and coming back in the same Zap card is there? Recap hardware: Sangoma A101 Echo cancel: yes (and zap show channel confirms it's enabled) I would think if echo cancel was the problem incoming faxes would fail as well? So...wtf?! I am surprised the 3660 is working outbound where the Sangoma is not, since it can receive fine. The 3660 has a HDV card in it with DSPs to do the processing but the load on the Asterisk servers barely goes above 0.00. So to recap: ATA works fine sending and my Asterisk servers are ok Sending the outbound call via SIP to my 3660 a hop away (DS3) to be routed out the PSTN (which then comes back to my Asterisk with the Sangoma card) works fine! Sending the outbound fax via the Zap channels on the Asterisk server (the same one that talks to the 3660 via SIP that works) FAILS Receiving faxes from anywhere into the Sangoma which talks to my 2nd asterisk server works fine as well! Bill -Original Message- From: Bill Gibbs Sent: Tue 1/2/2007 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: yet another faxing issue (outbound only, via ATA) 2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great! Never had an issue The problem, however is outbound. Sipura 1001 ATAs. Fax machine connected to the ATA. Registered to the 2nd asterisk box. Keep in mind this server runs voice calls just fine. Outbound calls from this box are ulaw The call is then sent via IAX2 (also tried SIP as well) to the Asterisk server w/ the PRI, then out to the world Hit and miss to send faxes out Echo cancellation is enabled on the PRI I have lowered the rxgain and txgain to -5.0, seems fine for voice. The ATA is running 3.1.8 firmware from Sipura with fax detect turned Usually the faxes fail, but sometimes you will get all the pages, but only a fraction of the page. I have tried turning off ECM but still the same issue. I would suspect the Sangom or IAX2, or something of that nature except receiving faxes traveling to the 2nd asterisk box works just fine! I also tried to register the ATA to the primary Asterisk server w/ the PRI, same exact issue. Any ideas - better luck w/ Grandstream? I suspect the problem is not Asterisk, or the Sangoma, or jitter or bandwidth since receiving faxes works fine. I did not try to receive faxes through the ATA to the machine itself, I tried that a few months ago during other testing, never got it to work so I never tried again once I got NVFaxDetect working for email. My next step is to connect the fax machine to a Wildcard X100P. Any other suggestions? Black magic? Voodoo? 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
Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
Do you know If its possible to do the same with Dock and Talk and an ATA GrandStream HandyTone 386? Thanks Joao Pereira Jonathan Attwood wrote: I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk. Because I'm using Asterisk, I cannot use voice dialling, however inbound outbound calls work extremely well. I have Asterisk outbound routes set up to make a calls to cell phones go through the Dock-n-Talk. On 1/1/06, *Brian McEntire* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Is anyone familiar with cell phone switches that allow routing cell phone calls through in-home wiring? One example of these devices is the Phone Labs Dock-N-Talk. It says it keeps your cell charged when you are home and connects your cell (for incoming and outgoing calls) to your home wiring or cordless phones. But it also has features such as allowing speed dialing and voice dialing from extensions if your cell phone has those features. So I'm not sure if the device offers a fully compatible FXO signalling. I'm currently running Asterisk with 1 POTS and 1 VOIP (via Sipura 3000) lines coming into Zaptel FXS modules, and then I have two FXO modules for two extensions. I'm thinking of doing away with the land line. Should something like the Dock-N-Talk allow substituting a cell phone line for the POTS line? ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] queues - limiting ringing calls to queue members
Hello, I'm using asterisk queues, for reception phone, and I have small problem: I have only one phone as queue member, and the problem is, that ALL channels waiting in queue are ringing on it. And if there are too many people ringing on it, it's not possible to use attended transfer then... Is it possible to limit maximum ringing calls from queue? or some other tip? thanks a lot in advance! best regards Nik ___ --Bandwidth 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] Best Hardware for Asterisk Server?
On Tue, 2 Jan 2007, Colin Anderson wrote: ASUS motherboards, in particular, have worked for me perfectly, everytime with both Digium and Sangoma cards. They are also easy to work with and well documented. I'd second that. I've been using Asus motherboards for over 10 years now in various LAMP type servers and desktop PCs. 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] vzaphfc?
On Fri, 29 Dec 2006, Julian J. M. wrote: It's not necessary to recompile the kernel for mISDN support. Check http://www.laimbock.com/asterisk/ Grab the mISDN source rpm, and build it. $ wget http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm $ rpmbuild --rebuild mISDN-cvs20061107-2_fc6.lc.src.rpm then check /usr/src/redhat/RPMS/i386/ You should have the kernel modules and userspace applications. Once installed, I could enable chan_misdn in asterisk 1.4 without issue, and it's working great in NT mode with ISDN phones. I haven't tested asterisk 1.2, but there is no it shouldn't work as well. I deleted all the bristuff modules i could find plus the old asterisk libs, compiled zaptel, libpri and asterisk from scratch but can't get it to work. First I get errors about something I guess is missing from misdn, later errors about zaptel. I'll just toss the HFC-S card and convert the ISDN line to analog. These are the errors : .mISDN_close: fid(14) isize(131072) inbuf(0x2a96ee6010) irp(0x2a96ee6010) iend(0x2a96ee6010) Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config: misdn.conf: ports=(null) (section: intern) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config: misdn.conf: ports=(null) (section: first_extern) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config: misdn.conf: ports=(null) (section: second_extern) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. P[ 0] Got: 1 from get_ports P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS EXPERIMENTAL!) ..Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown signalling method 'bri_cpe_ptmp' Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10228 setup_zap: Signalling must be specified before any channels are. Jan 2 23:07:23 WARNING[25747]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 2 23:07:23 WARNING[25747]: loader.c:554 load_modules: Loading module chan_zap.so failed! Cheers! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Admin manual for Linksys Sipura SPA-2102
Hi, Anyone knows where to get the admin (not the end user) manual for the linksys spa2102. This model is the 2 analog port+router. There are a lot of advanced options that I would like to see what they do. Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth 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] OnHook Call Announcement...
I have a customer that is asking for a feature called On Hook Call Announcement. The way he explains it is that when someone is on another call you can sort of break in into their conversation but only the local person hears you and not the external caller. Basically he wants to use this function so he can call anyone in the company even if they are already on a call (he is the big boss). I saw that there is a feature coming in 1.4 called Whisper paging that may do something like this but I need to know if it is possible to do it in 1.2 because there is still no support for Unicall on 1.4 -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chàvez Prats Director de Tecnologìa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extension problems
Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) I end up getting this when I call from 2000 to 2001. 2000, 2002, and 2001 all exist in sip.conf and I connect using them. I have all three setup to use the from-sip context. Any suggestions on what is happening? [from-sip] exten = 2999,1,VoicemailMain([EMAIL PROTECTED]) exten = 2000,1,Dial(SIP/2000,20) exten = 2000,2,Voicemail([EMAIL PROTECTED]) exten = 2000,3,PlayBack(vm-goodbye) exten = 2000,103,Hangup exten = 2001,1,Dial(SIP/2001) exten = 2001,2,Voicemail([EMAIL PROTECTED]) exten = 2001,3,PlayBack(vm-goodbye) exten = 2001,103,Hangup exten = 2002,1,Dial(SIP/2002) exten = 2002,103,Hangup ___ --Bandwidth 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] queues - limiting ringing calls to queue members
Nikola, Check the maxlen parameter for the queue... Also check the sample queues.conf distributed with Asterisk source, which somehow includes queue parameter documentation. If set, maxlen will limit the number of calls in the queue. Cheers, -- Ex Vito On 1/2/07, Nikola Ciprich [EMAIL PROTECTED] wrote: Hello, I'm using asterisk queues, for reception phone, and I have small problem: I have only one phone as queue member, and the problem is, that ALL channels waiting in queue are ringing on it. And if there are too many people ringing on it, it's not possible to use attended transfer then... Is it possible to limit maximum ringing calls from queue? or some other tip? thanks a lot in advance! best regards Nik ___ --Bandwidth 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] Double quotes in CDRUserField?
Question: I'm trying to put a double quote into the CDRUserField. What I end up with is a pair of double quotes. Example: exten = s,n,SetCDRUserField(data) exten = s,n,AppendCDRUserField() exten = s,n,AppendCDRUserField(moredata) My record will look like this: datamoredata What I want is: datamoredata The wiki mentions using a backslash in order to 'quote the character' as it says. However, this example: exten = s,n,SetCDRUserField(data) exten = s,n,AppendCDRUserField(\) exten = s,n,AppendCDRUserField(moredata) Yields the same results: datamoredata Is there something that I'm missing? Thanks, MC P.S. I'm using CSV for my CDR's ___ --Bandwidth 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] vzaphfc?
Remove from zapata.conf the lines re bristuff (bri_cpe_ptmp, etc).
Setup misdn:
/etc/init.d/misdn-init config
vi /etc/misdn-init.conf(check it's ok, NT or TE, PTP or PTMP...)
/etc/init.d/misdn-init start
chkconfig --add misdn-init
Setup chan_misdn, in /etc/asterisk/misdn.conf. At the end:
[telco]
port=1
context=from-pstn
msns=*
Then, in extensions.conf:
exten = _,1,Set(CALLERID(num)=00)
exten = _.,2,Dial(misdn/g:telco/${EXTEN})
Julian J. M.
On 1/2/07, Remco Barendse [EMAIL PROTECTED] wrote:
On Fri, 29 Dec 2006, Julian J. M. wrote:
It's not necessary to recompile the kernel for mISDN support. Check
http://www.laimbock.com/asterisk/
Grab the mISDN source rpm, and build it.
$ wget
http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm
$ rpmbuild --rebuild mISDN-cvs20061107-2_fc6.lc.src.rpm
then check /usr/src/redhat/RPMS/i386/
You should have the kernel modules and userspace applications. Once
installed, I could enable chan_misdn in asterisk 1.4 without issue,
and it's working great in NT mode with ISDN phones. I haven't tested
asterisk 1.2, but there is no it shouldn't work as well.
I deleted all the bristuff modules i could find plus the old asterisk
libs, compiled zaptel, libpri and asterisk from scratch but can't get it
to work.
First I get errors about something I guess is missing from misdn, later
errors about zaptel.
I'll just toss the HFC-S card and convert the ISDN line to analog.
These are the errors :
.mISDN_close: fid(14) isize(131072) inbuf(0x2a96ee6010) irp(0x2a96ee6010)
iend(0x2a96ee6010)
Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config:
misdn.conf: ports=(null) (section: intern) invalid or out of range.
Please edit your misdn.conf and then do a misdn reload.
Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config:
misdn.conf: ports=(null) (section: first_extern) invalid or out of
range. Please edit your misdn.conf and then do a misdn reload.
Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config:
misdn.conf: ports=(null) (section: second_extern) invalid or out of
range. Please edit your misdn.conf and then do a misdn reload.
P[ 0] Got: 1 from get_ports
P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS
EXPERIMENTAL!)
..Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown
signalling method 'bri_cpe_ptmp'
Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10228 setup_zap: Signalling must
be specified before any channels are.
Jan 2 23:07:23 WARNING[25747]: loader.c:414 __load_resource: chan_zap.so:
load_module failed, returning -1
Jan 2 23:07:23 WARNING[25747]: loader.c:554 load_modules: Loading module
chan_zap.so failed!
Cheers!
___
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Re: [asterisk-users] extension problems
Vulpes Velox wrote: Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) I end up getting this when I call from 2000 to 2001. 2000, 2002, and 2001 all exist in sip.conf and I connect using them. I have all three setup to use the from-sip context. Any suggestions on what is happening? Are you sure that 2001 is registered? ___ --Bandwidth 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] OnHook Call Announcement...
Its Called Off Hook Call Announcement And Asterisk In 1.2 Can Not Do This I Dont Know About 1.4 On 1/2/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a customer that is asking for a feature called On Hook Call Announcement. The way he explains it is that when someone is on another call you can sort of break in into their conversation but only the local person hears you and not the external caller. Basically he wants to use this function so he can call anyone in the company even if they are already on a call (he is the big boss). I saw that there is a feature coming in 1.4 called Whisper paging that may do something like this but I need to know if it is possible to do it in 1.2 because there is still no support for Unicall on 1.4 -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chàvez Prats Director de Tecnologìa +52-55-91169161 ext 2001 ___ --Bandwidth 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: yet another faxing issue (outbound only, via ATA)
On 1/2/07, Bill Gibbs [EMAIL PROTECTED] wrote: Echo cancel: yes (and zap show channel confirms it's enabled) I would think if echo cancel was the problem incoming faxes would fail as well? This is only a guess. The Sangoma is detecting the fax when it receives it, and is turning off echo cancel. However, when box b sends via IAX2 or SIP to box a, the Sangoma no longer knows that it is a fax transmission and is continuing echo cancellation. The Cisco 3660 recognizes that it is a fax and turns off echo (or doesn't have echo cancellation). Question: If you turn OFF echo cancellation, does it work then? ___ --Bandwidth 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: yet another faxing issue (outbound only, via ATA)
Haven't yet. Gotta wait until the calls stop flowing in/out. It's a production system. That's on the list of tonight. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Tuesday, January 02, 2007 8:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: yet another faxing issue (outbound only,via ATA) On 1/2/07, Bill Gibbs [EMAIL PROTECTED] wrote: Echo cancel: yes (and zap show channel confirms it's enabled) I would think if echo cancel was the problem incoming faxes would fail as well? This is only a guess. The Sangoma is detecting the fax when it receives it, and is turning off echo cancel. However, when box b sends via IAX2 or SIP to box a, the Sangoma no longer knows that it is a fax transmission and is continuing echo cancellation. The Cisco 3660 recognizes that it is a fax and turns off echo (or doesn't have echo cancellation). Question: If you turn OFF echo cancellation, does it work then? ___ --Bandwidth 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] Double quotes in CDRUserField?
Michael Collins wrote: Question: I'm trying to put a double quote into the CDRUserField. What I end up with is a pair of double quotes. Example: exten = s,n,SetCDRUserField(data) exten = s,n,AppendCDRUserField() exten = s,n,AppendCDRUserField(moredata) My record will look like this: datamoredata It's common for CSV files to escape quotes by putting two of them to indicate it is a quote within the string, not the end of the string. Perhaps you could accomplish what you're going for with something else, say an underscore character? Regards, Trevor Peirce ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [A*UG] How to show a debugging remark in a sip or extensions context?
Chris how would I use 'verbose' in a dialplan context? A sample line? Larry Chris Tooley wrote: If you mean in the dialplan, you can use NoOp or verbose (verbose being something that will get logged too), and if you mean in the asterisk code, there are logging examples all over the place. -Original Message- From: [EMAIL PROTECTED] on behalf of Larry Alkoff Sent: Tue 1/2/2007 11:22 AM To: Asterisk-users; Austin-asterisk-users Subject: [A*UG] How to show a debugging remark in a sip or extensions context? I would like to show a remark that would show call progress and appear on the CLI screen. The remark should be in the code of a sip [channel] or extentions [context] If I can't send my own remark, what little used 'show' command could I insert in the code? Can this be done? -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth 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] connecting asterisk (trixbox) to traditional phone lines?
Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OnHook Call Announcement...
Its Called Off Hook Call Announcement And Asterisk In 1.2 Can Not Do This I Dont Know About 1.4 This would indeed be off-hook announcement - doesn't call waiting use this feature already? Can't see much difference from the description - even the big boss won't like forced barge-in if his horses and men are talking to a client. Yuan Liu On 1/2/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a customer that is asking for a feature called On Hook Call Announcement. The way he explains it is that when someone is on another call you can sort of break in into their conversation but only the local person hears you and not the external caller. Basically he wants to use this function so he can call anyone in the company even if they are already on a call (he is the big boss). I saw that there is a feature coming in 1.4 called Whisper paging that may do something like this but I need to know if it is possible to do it in 1.2 because there is still no support for Unicall on 1.4 -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chàvez Prats Director de Tecnologìa +52-55-91169161 ext 2001 ___ --Bandwidth 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] Error compiling chan_vpb
DiegoF wrote: chan_vpb.o:chan_vpb.cc:(.text+0x4da6): first defined here /usr/bin/ld: Warning: size of symbol `load_module' changed from 3274 in chan_vpb.o to 3926 in chan_vpb.oo collect2: ld devolvi el estado de salida 1 make[1]: *** [chan_vpb.so] Error 1 rm chan_vpb.o make: *** [channels] Error 2 hello, if somebody knows like solving this error, to him it will be been thankful. This has been fixed in Subversion branch-1.4; the fix will be included in the Asterisk 1.4.1 release. ___ --Bandwidth 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] connecting asterisk (trixbox) to traditional phone lines?
On Jan 2, 2007, at 10:15 PM, blackwater dev wrote: I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. I just went thru this recently. I ended up buying a compatible modem on Ebay. You can find them easily if you search for FXO or X100 but then you may also end up paying a premium to get one that is specifically being sold to the Asterisk community. (keep in mind premium being around $30, so we still aren't talking about an outrageous price) What I did was checked the voip-info.org wiki on modem based FXOs and then searched ebay for modems listed with the correct chipsets. I lucked out and found one for $2.00 (with shipping I think it cost me $8.00 total). Mine is shows up as a Motorola X100 (or something to that effect). Seems to work fine, although I wasn't able to get Caller-ID working correctly (but I think that was a settings issue and I stopped pursuing it as it wasn't important for my pitching Asterisk). I too did this using Trixbox. -chris www.mythtech.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] connecting asterisk (trixbox) to traditional phonelines?
From: blackwater dev [EMAIL PROTECTED] Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. Thanks! Add an FXO to the box for about $18. (An X100P clone could be cheaper.) You can also go SIPphone/Gizmo and or another provider for $5/month but it's probably more troublesome and can cost more. 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] connecting asterisk (trixbox) to traditional phone lines?
To go nice and cheaply, you could just get a free number from IPKALL.com or Stanaphone.com.. And do it all over IP... -t- On Jan 2, 2007, at 10:15 PM, blackwater dev wrote: Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Buying
Your best option will be to contact a local Asterisk integrator - and get them started on the work. PaulH On Tue, 2007-01-02 at 12:12 -0800, Khaled wrote: Dear Guys Merry Christmas and happy new year . Please do any one knows from where I can buy a full pbx corporate cd and integrated with exchange server and life communication server . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth 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] connecting asterisk (trixbox) to traditional phone lines?
I agree with this... The cheapest way is to do this without anymore hardware. Grab a pay-as-you-go VoIP provider (VoIPJet, Unlmitel, Gizmo Project, etc.) and setup a trunk. They'll give you a number callable from the PSTN, and that's all you need. The setup you have already can handle a voip trunk with no additional hardware. Ales On 1/2/07, Todd H [EMAIL PROTECTED] wrote: To go nice and cheaply, you could just get a free number from IPKALL.com or Stanaphone.com.. And do it all over IP... -t- On Jan 2, 2007, at 10:15 PM, blackwater dev wrote: Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vzaphfc?
On Tue, Jan 02, 2007 at 11:10:56PM +0100, Remco Barendse wrote: On Fri, 29 Dec 2006, Julian J. M. wrote: It's not necessary to recompile the kernel for mISDN support. Check http://www.laimbock.com/asterisk/ Grab the mISDN source rpm, and build it. $ wget http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm $ rpmbuild --rebuild mISDN-cvs20061107-2_fc6.lc.src.rpm then check /usr/src/redhat/RPMS/i386/ You should have the kernel modules and userspace applications. Once installed, I could enable chan_misdn in asterisk 1.4 without issue, and it's working great in NT mode with ISDN phones. I haven't tested asterisk 1.2, but there is no it shouldn't work as well. I deleted all the bristuff modules i could find plus the old asterisk libs, compiled zaptel, libpri and asterisk from scratch but can't get it to work. First I get errors about something I guess is missing from misdn, later errors about zaptel. I'll just toss the HFC-S card and convert the ISDN line to analog. These are the errors : .mISDN_close: fid(14) isize(131072) inbuf(0x2a96ee6010) irp(0x2a96ee6010) iend(0x2a96ee6010) Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config: misdn.conf: ports=(null) (section: intern) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config: misdn.conf: ports=(null) (section: first_extern) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. Jan 2 23:07:23 WARNING[25747]: misdn_config.c:642 _build_port_config: misdn.conf: ports=(null) (section: second_extern) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. P[ 0] Got: 1 from get_ports P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS EXPERIMENTAL!) ..Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown signalling method 'bri_cpe_ptmp' our Asterisk is not bristuffed. And you don't expect to use ZapBRI, anyway. BTW: with 1.4 and latest 1.2, bristuffed zaptel could basicaly work with the signalling type pri_cpe/pri_net, though this is not well-tested and may not perform as well as bristuffed asterisk/libpri. Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10228 setup_zap: Signalling must be specified before any channels are. Jan 2 23:07:23 WARNING[25747]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 2 23:07:23 WARNING[25747]: loader.c:554 load_modules: Loading module chan_zap.so failed! You have zaptel channels configured in your zapata.conf . BTW: I guess that this is 1.2 and not 1.4, because 1.4 should not fail just becasue chan_zap.so has failed to set up some channels. -- 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
Re: [asterisk-users] connecting asterisk (trixbox) to traditional phone lines?
On Tue, Jan 02, 2007 at 10:25:45PM -0500, cb wrote: On Jan 2, 2007, at 10:15 PM, blackwater dev wrote: I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. I just went thru this recently. I ended up buying a compatible modem on Ebay. You can find them easily if you search for FXO or X100 but then you may also end up paying a premium to get one that is specifically being sold to the Asterisk community. (keep in mind premium being around $30, so we still aren't talking about an outrageous price) Those 30$ cards are as good as the 10$ cards. Same low quality. They are nice for playing games. If you're lucky enough it may actually work for you. In the worst case you only lost 30$ ... What I did was checked the voip-info.org wiki on modem based FXOs and then searched ebay for modems listed with the correct chipsets. I lucked out and found one for $2.00 (with shipping I think it cost me $8.00 total). Mine is shows up as a Motorola X100 (or something to that effect). Seems to work fine, although I wasn't able to get Caller-ID working correctly (but I think that was a settings issue and I stopped pursuing it as it wasn't important for my pitching Asterisk). I don't recall any special issues with caller ID with X100P. -- 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
RE: [asterisk-users] Double quotes in CDRUserField?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Trevor Peirce Sent: Tuesday, January 02, 2007 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Double quotes in CDRUserField? Michael Collins wrote: Question: I'm trying to put a double quote into the CDRUserField. What I end up with is a pair of double quotes. Example: exten = s,n,SetCDRUserField(data) exten = s,n,AppendCDRUserField() exten = s,n,AppendCDRUserField(moredata) My record will look like this: datamoredata It's common for CSV files to escape quotes by putting two of them to indicate it is a quote within the string, not the end of the string. Perhaps you could accomplish what you're going for with something else, say an underscore character? Regards, Trevor Peirce Under the circumstances I think that is the easiest thing to do. I can do some minor shell scripting to handle the parsing of the userfield. Thanks for the suggestion. -MC ___ --Bandwidth 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] vzaphfc?
On Wed, 3 Jan 2007, Tzafrir Cohen wrote: P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS EXPERIMENTAL!) ..Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown signalling method 'bri_cpe_ptmp' our Asterisk is not bristuffed. And you don't expect to use ZapBRI, anyway. BTW: with 1.4 and latest 1.2, bristuffed zaptel could basicaly work with the signalling type pri_cpe/pri_net, though this is not well-tested and may not perform as well as bristuffed asterisk/libpri. Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10228 setup_zap: Signalling must be specified before any channels are. Jan 2 23:07:23 WARNING[25747]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 2 23:07:23 WARNING[25747]: loader.c:554 load_modules: Loading module chan_zap.so failed! You have zaptel channels configured in your zapata.conf . So I should leave both zaptel.conf and zapata.conf completely empty? BTW: I guess that this is 1.2 and not 1.4, because 1.4 should not fail just becasue chan_zap.so has failed to set up some channels. This is Asterisk 1.2 indeed, i need the chan-sccp driver from Sergio Chersovani because it supports multiple phone registrations from 1 ip address and AFAIK that channel doesn't work on 1.4 yet. Is there a lot of difference in misdn support between 1.2 - 1.4? I tried googling for example configs with misdn but couldn't find any. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dialed Number missing from the CDR when using callfiles.
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Richard Lyman Sent: Saturday, December 30, 2006 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialed Number missing from the CDR when usingcallfiles. *snipped Second, when using a .call file (or the manager interface's Originate action) the 'Dial' action is executed BEFORE entry into the dialplan, so if it fails, nothing in your dialplan is executed and you get a somewhat *snipped not *exactly* true. you need to add ;this extension MUST be here for OriginateFailure triggers exten = failed,1,Hangup to your context used for *send too after connect* The one caveat here is that * actually cuts two CDR's for the call. This isn't normally a problem unless half the data you want is in CDR one and half is in the other! :) I have done some scripting to extract the relevant data from each record and condense it back down to one - a small price to pay to have the functionality that I really need. -MC ___ --Bandwidth 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] connecting asterisk (trixbox) to traditional phone lines?
try http://www.digitnetworks.com/X100P_FXO_PCI_Card_p/x100p.htm On Wed, 3 Jan 2007, Tzafrir Cohen wrote: On Tue, Jan 02, 2007 at 10:25:45PM -0500, cb wrote: On Jan 2, 2007, at 10:15 PM, blackwater dev wrote: I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. I just went thru this recently. I ended up buying a compatible modem on Ebay. You can find them easily if you search for FXO or X100 but then you may also end up paying a premium to get one that is specifically being sold to the Asterisk community. (keep in mind premium being around $30, so we still aren't talking about an outrageous price) Those 30$ cards are as good as the 10$ cards. Same low quality. They are nice for playing games. If you're lucky enough it may actually work for you. In the worst case you only lost 30$ ... What I did was checked the voip-info.org wiki on modem based FXOs and then searched ebay for modems listed with the correct chipsets. I lucked out and found one for $2.00 (with shipping I think it cost me $8.00 total). Mine is shows up as a Motorola X100 (or something to that effect). Seems to work fine, although I wasn't able to get Caller-ID working correctly (but I think that was a settings issue and I stopped pursuing it as it wasn't important for my pitching Asterisk). I don't recall any special issues with caller ID with X100P. -- 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 -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
