Re: [asterisk-users] Passing call duration to an AGI Script
Hi, What I did is first to dig a bit into the app_dial.c. I saw how the ANSWEREDTIME variable is created (end_time - answer_time). Then I added some lines to export the answer_time variable as a channel variable. I added these lines right after the answer_time decleration (line 1426 in ver 1.4.4) compiled and replaced the module. char toast2[80]; snprintf(toast2, sizeof(toast2), %ld, (long)(answer_time)); pbx_builtin_setvar_helper(chan, ANSWERTIME, toast2); This will put the call start time in unix timestamp in the channel variable ANSWERTIME. That's all. Hope it's helping. Adi. On 6/1/07, Luis Morales [EMAIL PROTECTED] wrote: Hi Adi, My be better if you send us the code about how did you do to catch and retrive the data from asterisk. Regards, Luis Morales On Fri, 2007-06-01 at 01:21 +0300, Adi Simon wrote: Hi Martin, Thanks for your reply. Maybe I wasn't clear enough. I am already running AGI periodically inside a call and it runs just fine. I'm using a patch for asterisk (can be found here) to do so. In short i'm using it for a prepaid system that needs to allow more than one prepaid call to run simultaneously. Anyway, I solved my problem by changing the code a bit. I added an AGI variable that holds the timestamp of the call answer time, thus allowing me to use it as an anchor for knowing how much time passed since the beginning of the call. Thanks again, Adi. On 5/31/07, Martin Smith [EMAIL PROTECTED] wrote: Hi Adi, AGI is probably best viewed like any other dialplan application (and with DeadAGI something that happens after, but anyway) -- in my opinion. I've seen people do some pretty wild stuff with it, but in the end, when I wonder if the Manager interface or AGI interface is most appropriate for a given task, I ask questions like Would I want to do this with another application? Is this even possible with another application?. In your case, I'd say you probably couldn't say... periodically execute a dialplan application that runs in the middle of a call without interrupting the call (with AGI, anyway). I'd recommend using the Manager interface and polling for call durations / listening for events and acting on the information you get back (I'd assume the answered duration is one of those values you could poll for). Hope this helps -- others, please jump in if I'm way wrong :) Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 __ From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Adi Simon Sent: Thursday, May 31, 2007 5:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Passing call duration to an AGI Script Hi, I'm trying to find a way of passing the actual call duration (something like ANSWEREDTIME) to an AGI script that runs periodically during a call. Any ideas? Thanks, Adi. ___ --Bandwidth 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 -- .-.-.-.-.-.-.-.-.-.-.-.-.--.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-. Sigma Dental Plan Jefe de Soporte y Sistemas Telf. Oficina : +58(212)2646811 Cel.: +58(416)4242091 Caracas, Venezuela .-.-.-.-.-.-.-.-.-.-.-.-.--.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Crash
Hi I've two boxes connected via IAX2 Trunk were working fine from few days suddenly today one box is got crashed with this message 2007-06-03 12:25:37 WARNING[26511]: chan_sip.c:2612 sip_write: Can't send 4113608 type frames with SIP write my version of * is 1.2.14 on FC4 thanks arun ___ --Bandwidth 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] really strange behavior
In short, the 's' extension is not a catch-all. The use of 's' can be confusing. The best example I have of the use of 's' is when a ZAP call comes in on an analog line. IIRC, the book says something to the effect that 's' is for when, upon arrival in a context, the call has no other place to go. Works for 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
[asterisk-users] Asterisk Queue
HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Dial Problem
Hi,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
No problem with Auto Call
exten = _01N.,1,Dial(Zap/g1/${EXTEN},20)
the problem with zap channel
try callprogress with yes in zapata.conf
It may cause another problem, after remote party has picked up the
call and asterisk still does not know it. and in ringing status.
if your dial plan work fine now, then no need to change rxgain.
otherwise.
Just Increase your rxgain value. try with different values and choose
best one.
if rxgain greater then desired value ?? you my receive invalid report
that remote party has picked up.
if rxgain less then desired value ?? you my receive invalid ringing
report after call is answered.
so adjust it according your requirement and also check noise and quality
your PSTN lines.
Regards
Nasir Iqbal
ICT Innovations
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Re: [asterisk-users] Auto Dial Problem
Hi,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
No problem with Auto Call
exten = _01N.,1,Dial(Zap/g1/${EXTEN},20)
the problem with zap channel
try callprogress with yes in zapata.conf
It may cause another problem, after remote party has picked up the
call and asterisk still does not know it. and in ringing status.
if your dial plan work fine now, then no need to change rxgain.
otherwise.
Just Increase your rxgain value. try with different values and choose
best one.
if rxgain greater then desired value ?? you my receive invalid report
that remote party has picked up.
if rxgain less then desired value ?? you my receive invalid ringing
report after call is answered.
so adjust it according your requirement and also check noise and quality
your PSTN lines.
Regards
Nasir Iqbal
ICT Innovations
___
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Re: [asterisk-users] Asterisk Queue
And you don't find that sufficiently self-explanatory? On Sun, 2007-06-03 at 13:02 +0400, Arun Kumar wrote: HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Cheers, Mattt. - ROMATel - VoIP made easy - http://romatel.net - SpotSafe - WiFi Hotspot solution - http://spotsafe.net There are only 10 kinds of people. Those who understand binary, and those that don't... ___ --Bandwidth 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 Queue
Hi my * box is giving me these warning and b'coz of second warning line my agents are not able to hear the announcement in the queue some time it happen many time 2007-06-03 13:40:30 WARNING[28016]: chan_sip.c:2612 sip_write: Can't send 4113568 type frames with SIP write 2007-06-03 13:40:30 WARNING[28016]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun ___ --Bandwidth 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 Queue
Hi sorry for not details. when ever I see this message on * console my agents are not able to listen to announcement. thanks arun On 6/3/07, Mattt [EMAIL PROTECTED] wrote: And you don't find that sufficiently self-explanatory? On Sun, 2007-06-03 at 13:02 +0400, Arun Kumar wrote: HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use http://lists.digium.com/mailman/listinfo/asterisk-usersrs Cheers, Mattt. - ROMATel - VoIP made easy - http://romatel.net - SpotSafe - WiFi Hotspot solution - http://spotsafe.net There are only 10 kinds of people. Those who understand binary, and those that don't... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to make busy sign
Hi, Steve Totaro írta: It sounds like the Meridian GSM devices connect via analog line so if you connected the ports on the Meridian currently terminating into the GSM devices go into asterisk via a four port FXS (guessing the GSM devices supply dialtone) then the routing in the Meridian would not change and your calls would go to the asterisk machine. Yeah good idea, ty! :) thanks, tsabi ___ --Bandwidth 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] Telefonica in Czech Republic Blocking VOIP ?
Hi, Does any in the Czech republic had an issue with Telifonica blocking port 5060 ? I have a client there that is unable to register with our servers in the US. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] linksys wip300 firmware
If your getting an error about invalid Octet key or something make sure you are doing the load from a Windows Computer it's been noted that a lot of people have had problems loading that version from various OS's. I went through a few computers before it took the load (for the record it worked from winxp / ie7) On Jun 2, 2007, at 7:48 PM, Ilan Rabinovitch wrote: Hello, Does anyone have access to version 1.00.07 of the Linksys WIP 300 firmware? The only version on their site at the moment is 1.00.09, which the phone refuses to load. Regards, Ilan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth 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] System Application, Fail/Timeout Issue
Thank you. Seems the rtptimeout did the trick. Sphinx2 doesn't always manage to work out what i'm saying but when it does, everything works. Thanks again. - Email sent from www.virginmedia.com/email Virus-checked using McAfee(R) Software and scanned for spam ___ --Bandwidth 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] Centos kernel source
On Sun, Jun 03, 2007 at 02:50:17AM -0700, Khaled Chehab wrote: I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2 ... error appears From where I can get the missing rpms .or kernel source you need kernel-devel or kernel-smp-devel, depending on your running kernel version. -- 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
[asterisk-users] zaptel on CENTOS servercd
I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there is someone did that, Please in need to have the installation procedure step by step. Its too urgent for me . Thanks alot 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] Centos kernel source
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Sunday, June 03, 2007 8:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Centos kernel source On Sun, Jun 03, 2007 at 02:50:17AM -0700, Khaled Chehab wrote: I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2 ... error appears From where I can get the missing rpms .or kernel source you need kernel-devel or kernel-smp-devel, depending on your running kernel version. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Try yum install kernel-devel, yum update. Thanks, Steve Totaro ___ --Bandwidth 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] zaptel on CENTOS servercd
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Sunday, June 03, 2007 6:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: [asterisk-users] zaptel on CENTOS servercd I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there is someone did that, Please in need to have the installation procedure step by step. Its too urgent for me . Thanks alot Regards Stop the suffering! Just use the Trixbox install CD and then remove all things Trixbox. IE Startup scripts, configs, database, whatever other stuff is on there. That is of course if you don't want a full featured GUI PBX and want a more pure version of Asterisk. Thanks, Steve ___ --Bandwidth 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] zaptel on CENTOS servercd
On Sun, Jun 03, 2007 at 09:10:19AM -0400, Steve Totaro wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Sunday, June 03, 2007 6:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: [asterisk-users] zaptel on CENTOS servercd I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there is someone did that, Please in need to have the installation procedure step by step. Its too urgent for me . Thanks alot Regards Stop the suffering! Just use the Trixbox install CD and then remove all things Trixbox. IE Startup scripts, configs, database, whatever other stuff is on there. That is of course if you don't want a full featured GUI PBX and want a more pure version of Asterisk. And an obsolete centos kernel with a cripelling yum configuration to prevent you from upgrading kernel. No thanks you. Also note that if you remove startup scripts, you break install of zaptel... -- 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] Centos kernel source
Steve Totaro wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Sunday, June 03, 2007 8:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Centos kernel source On Sun, Jun 03, 2007 at 02:50:17AM -0700, Khaled Chehab wrote: I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2 ... error appears From where I can get the missing rpms .or kernel source you need kernel-devel or kernel-smp-devel, depending on your running kernel version. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Try yum install kernel-devel, yum update. Here's what I did: yum update reboot yum install kernel-devel-`uname -r` yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool make automake automake14 automake15 automake16 automake17 \ bison byacc flex libtermcap libtermcap-devel newt newt-devel \ ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel cd /usr/src/ wget -c http://ftp.digium.com/pub/zaptel/releases/zaptel-1.4.1.tar.gz tar -xzf zaptel-1.4.1.tar.gz cd /usr/src/zaptel-1.4.1/ ./configure make clean make make install make config modprobe ztdummy Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [asterisk-users] Centos kernel source
I already did what you said,please see the log results in zaptel.rar attached when I compile zapltel using make * 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. * zaptel.rar Description: Binary data ___ --Bandwidth 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] zaptel on CENTOS servercd
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Sunday, June 03, 2007 9:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] zaptel on CENTOS servercd On Sun, Jun 03, 2007 at 09:10:19AM -0400, Steve Totaro wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Sunday, June 03, 2007 6:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: [asterisk-users] zaptel on CENTOS servercd I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there is someone did that, Please in need to have the installation procedure step by step. Its too urgent for me . Thanks alot Regards Stop the suffering! Just use the Trixbox install CD and then remove all things Trixbox. IE Startup scripts, configs, database, whatever other stuff is on there. That is of course if you don't want a full featured GUI PBX and want a more pure version of Asterisk. And an obsolete centos kernel with a cripelling yum configuration to prevent you from upgrading kernel. No thanks you. Also note that if you remove startup scripts, you break install of zaptel... -- Tzafrir Cohen ob*so*lete adj. 1. No longer in use: an obsolete word. 2. Outmoded in design, style, or construction: an obsolete locomotive. 3. Biology Vestigial or imperfectly developed, especially in comparison with other individuals or related species; not clearly marked or seen; indistinct. Used of an organ or other part of an animal or plant. Maybe, but I don't think the definition is an accurate adjective for the stock kernel. At any rate, simply removing the commented lines concerning updating the kernel in the repo files fixes that problem of upgrading the cripelling yum configuration. That should take you less than one minute. Then downloading and building and installing Asterisk in the regular fashion fixes any zaptel issues. Takes a several minutes depending on the machine you are compiling and installing on. Not the big deal that Cohen makes it out to be. Thank you, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [asterisk-users] Centos kernel source
I already did what you said,please see the results when you compile using make ptel-1.4.1/pciradio.c:1139: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1534: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1535: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1538: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1541: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1544: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1547: warning: implicit declaration of function `outl' /usr/src/zaptel-1.4.1/pciradio.c:1547: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1547: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1548: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1548: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1549: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1549: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1551: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1551: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1552: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1552: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1553: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1553: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1556: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1561: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1573: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c: In function `pciradio_enable_interrupts': /usr/src/zaptel-1.4.1/pciradio.c:1584: error: dereferencing pointer to /usr/src/zaptel-1.4.1/pciradio.c:1683: warning: implicit declaration of function `pci_free_consistent' /usr/src/zaptel-1.4.1/pciradio.c:1683: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1683: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1698: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1699: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1701: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1702: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1703: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1704: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1759: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1760: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1763: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1766: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1769: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c: At top level: /usr/src/zaptel-1.4.1/pciradio.c:1773: error: elements of array `pciradio_pci_tbl' have incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1775: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1775: warning: (near initialization for `pciradio_pci_tbl[1]')
RE: [asterisk-users] Centos kernel source
It would be good if you did not trip the part of I already did what you said. I am not sure what you did. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Sunday, June 03, 2007 8:03 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: FW: [asterisk-users] Centos kernel source I already did what you said,please see the log results in zaptel.rar attached when I compile zapltel using make * 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] Centos kernel source
You already did what WHO said? What did you already do? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Sunday, June 03, 2007 8:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: FW: [asterisk-users] Centos kernel source I already did what you said,please see the results when you compile using make ptel-1.4.1/pciradio.c:1139: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1534: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1535: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1538: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1541: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1544: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1547: warning: implicit declaration of function `outl' /usr/src/zaptel-1.4.1/pciradio.c:1547: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1547: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1548: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1548: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1549: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1549: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1551: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1551: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1552: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1552: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1553: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1553: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1556: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1561: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1573: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c: In function `pciradio_enable_interrupts': /usr/src/zaptel-1.4.1/pciradio.c:1584: error: dereferencing pointer to /usr/src/zaptel-1.4.1/pciradio.c:1683: warning: implicit declaration of function `pci_free_consistent' /usr/src/zaptel-1.4.1/pciradio.c:1683: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1683: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1698: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1699: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1701: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1702: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1703: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1704: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1759: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1760: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1763: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1766: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1769: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c: At top level: /usr/src/zaptel-1.4.1/pciradio.c:1773: error: elements of array `pciradio_pci_tbl' have incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: excess elements in struct initializer /usr/src/zaptel-1.4.1/pciradio.c:1774: warning: (near initialization for `pciradio_pci_tbl[0]') /usr/src/zaptel-1.4.1/pciradio.c:1774:
Re: [asterisk-users] zaptel on CENTOS servercd
On Sun, Jun 03, 2007 at 10:14:10AM -0400, Steve Totaro wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Sunday, June 03, 2007 9:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] zaptel on CENTOS servercd On Sun, Jun 03, 2007 at 09:10:19AM -0400, Steve Totaro wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Sunday, June 03, 2007 6:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: [asterisk-users] zaptel on CENTOS servercd I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there is someone did that, Please in need to have the installation procedure step by step. Its too urgent for me . Thanks alot Regards Stop the suffering! Just use the Trixbox install CD and then remove all things Trixbox. IE Startup scripts, configs, database, whatever other stuff is on there. That is of course if you don't want a full featured GUI PBX and want a more pure version of Asterisk. And an obsolete centos kernel with a cripelling yum configuration to prevent you from upgrading kernel. No thanks you. Also note that if you remove startup scripts, you break install of zaptel... -- Tzafrir Cohen ob*so*lete adj. 1. No longer in use: an obsolete word. 2. Outmoded in design, style, or construction: an obsolete locomotive. 3. Biology Vestigial or imperfectly developed, especially in comparison with other individuals or related species; not clearly marked or seen; indistinct. Used of an organ or other part of an animal or plant. Obsolete as in buggy. Specifically, I have asked our dear friend jbot for its opinion: jbot well, centosbug is a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h Now, if you wish that to beginners, then all the best for you. Maybe, but I don't think the definition is an accurate adjective for the stock kernel. At any rate, simply removing the commented lines concerning updating the kernel in the repo files fixes that problem of upgrading the cripelling yum configuration. That should take you less than one minute. Then downloading and building and installing Asterisk in the regular fashion fixes any zaptel issues. And is that supposed to save you the simple: yum install kernel-devel Because it sounds like you just did that. And a whole lot more. Enjoy :-) And then again, with Debian it is even simpler ;-) -- 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: FW: [asterisk-users] Centos kernel source
On Sun, Jun 03, 2007 at 05:07:49AM -0700, Khaled Chehab wrote: I already did what you said,please see the results when you compile using make ptel-1.4.1/pciradio.c:1139: error: dereferencing pointer to incomplete type /usr/src/zaptel-1.4.1/pciradio.c:1534: error: dereferencing pointer to incomplete type Any specific reason you don't use zaptel 1.2.4.1 (latest version)? -- 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] Centos kernel source
Thanks for the detailed procedure. One update: On Sun, Jun 03, 2007 at 03:31:53PM +0200, Philipp Kempgen wrote: Here's what I did: yum update reboot yum install kernel-devel-`uname -r` yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool make automake automake14 automake15 automake16 automake17 \ bison byacc flex libtermcap libtermcap-devel newt newt-devel \ ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel cd /usr/src/ wget -c http://ftp.digium.com/pub/zaptel/releases/zaptel-1.4.1.tar.gz tar -xzf zaptel-1.4.1.tar.gz cd /usr/src/zaptel-1.4.1/ There is a later version: # if you get an 404 message here, check if there's a newer version of # zaptel: wget http://ftp.digium.com/pub/zaptel/zaptel-1.4.2.1.tar.gz tar -xzf zaptel-1.4.2.1.tar.gz cd zaptel-1.4.2.1 #and from here continue as usual: ./configure make clean make make install make config modprobe ztdummy -- 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] really strange behavior
Understood, it is not the catch all but, what if I am designing a server that needs to accept calls from 15 or more 1800 numbers? How would you now channel it to a catch all? Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Sunday, June 03, 2007 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] really strange behavior In short, the 's' extension is not a catch-all. The use of 's' can be confusing. The best example I have of the use of 's' is when a ZAP call comes in on an analog line. IIRC, the book says something to the effect that 's' is for when, upon arrival in a context, the call has no other place to go. Works for 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] Centos kernel source
Tzafrir Cohen wrote: Thanks for the detailed procedure. One update: On Sun, Jun 03, 2007 at 03:31:53PM +0200, Philipp Kempgen wrote: Here's what I did: yum update reboot yum install kernel-devel-`uname -r` yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool make automake automake14 automake15 automake16 automake17 \ bison byacc flex libtermcap libtermcap-devel newt newt-devel \ ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel cd /usr/src/ wget -c http://ftp.digium.com/pub/zaptel/releases/zaptel-1.4.1.tar.gz tar -xzf zaptel-1.4.1.tar.gz cd /usr/src/zaptel-1.4.1/ There is a later version: # if you get an 404 message here, check if there's a newer version of # zaptel: wget http://ftp.digium.com/pub/zaptel/zaptel-1.4.2.1.tar.gz tar -xzf zaptel-1.4.2.1.tar.gz cd zaptel-1.4.2.1 #and from here continue as usual: ./configure make clean make make install make config modprobe ztdummy That was just a copy paste from what I did on a CentOS 4.4 ServerCD several months ago. Not sure if all of the packages are needed for zaptel but it works for me. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reset Polycom phones remotely
Rob Schall wrote: Are you able to access the phone via a web browser? And did asterisk register the phone? If both are true and you set the always reboot flag to 1, then rebooted the phone by hand, there shouldn't be anything standing in the way. It seems that it will only reboot if certain .cfg files are changed. I did that, rebooted the phone, and now it reboots every time I send a check-cfg. -Stephen- ___ --Bandwidth 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 auto dial does not wait for answer
Hi:
[EMAIL PROTECTED] wrote:
Hi All,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
This will not work with this channel driver. Explanation follows.
My sample call file :
hannel: local/[EMAIL PROTECTED]
MaxRetries: 5
RetryTime: 300
WaitTime: 40
Account: Reminder
context: remindem
extension: s
priority: 1
Set: MSG=0135.20070601.0124787924
Set: APPTDT=20070601
Set: APPTTIME=0135
Set: APPTPHONE=0124787924
Set: CALLATTEMPTS=5
Set: CALLDELAY=300
My outbound-reminder context:
[outbound-reminder]
exten = _01N.,1,Dial(Zap/g1/${EXTEN},20)
Zaptel considers the call answered the moment it is bridged. You need
call progress detection of some kind. Are you using an analog card?
-Stephen-
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Re: [asterisk-users] Thank you Asterisk mailing list!
Ricardo Martins wrote: Thinking this way, I invite those who think about the open source communities just as a zero price and its mailing lists as a space to wait passively for answers, to rethink its own ideas. Before asking for something and adding trash to communities mailing lists, DO A WORK OF RESEARCH and then, send to the list what you made to solve (the solutions), even if you don't need the help of the list for that issue anymore. I often find solutions while writing out requests for assistance from the mailing list. Sometimes just structuring the problem and writing it out make the solution obvious. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP inbound/outbound connection taking too long
Gordon Henderson wrote: On Fri, 1 Jun 2007, Gavin Henry wrote: Dear all, I think this is common, or at least how it is supposed to be, but whening dialing over a ZAP channel, it's taking around 5~ seconds to ring on the over end, likewise inbound. This is just with a normal Dial command. It's normal for an analogue Zap channel. Not exactly. Explanation follows. Asterisk has to sieze the line (after a basic check to make sure the channel is free), that may entail a delay of a second or so while it makes sure there there is a dial-tone (actually, I'm not sure it waits for a dial-tone) Zap does not do dial tone detection. , then it sends the digits out via DTMF - that might take a second or 2 for a long number - then it's up to the PSTN switch at the other end to connect the call - depending on the technology, this might take several seconds. This depends on multiple factors -- you can tweak your DTMF output to speed up dialing. You'll have to use some trial and error to confirm that the PSTN is receiving the DTMF properly. I have another theory, though -- are you dialing from a SIP extension? The biggest delay, in my experience, is the time it takes for the SIP extension to recognize that a full number has been dialed. It doesn't pass the number to Asterisk until it has interpreted it as a complete number. Many SIP devices let you configure this behaviour. You should also compare how long it takes to complete a call from an analog extension that is directly on the line, so that you have a baseline for reference. You will also find that many key systems introduce delay like this. In any case, this is all tweakable to make it faster. -Stephen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan_mobile issue
Hello, I just did a fresh svn install of 1.4 trunk everything. Everything compiles and installs just fine. When I get to asterisk-addons, I cannot select chan_mobile in menuselect. Chan_mobile is not even an option in menuselect for asterisk trunk. I tried the latest patch which failed in many places but did add an option for chan_mobile in menuselect for asterisk but it still cannot be enabled. If chan_mobile is in the asterisk-addons trunk then I must just be missing something on where to enable it, right? The readme says nothing. This box is fedora core 6 with all the bluez stuff installed and loaded and a dongle attached. I can see and pair with the box with my cell phone so Bluetooth is working in linux. Ideas? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth 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] Loud noise instead of MOH
Hi Everybody, I'm experiencing this kind of issue. One ASTERISK 1.2.17 is connected to a Bristuffed HFC single ISDN channel card. Everything seems to work but sometimes the third party caller when listening to MOH listens some SSH! instead of MOH, this is not continuos, MOH plays ok for, say, 20 seconds then the sound and then another 30 seconds of good MOH. We have some SIP phones for extensions (Grandstream 496 ATA and One SPX2000). Sometimes also ATAs hung, not really got ATAs at all. Does Anybody have some hint? Ciao a tutti Mauro _ Don't just search. Find. Check out the new MSN Search! http://search.msn.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] really strange behavior
exten = _X.,1, On 6/3/07, BSumrall [EMAIL PROTECTED] wrote: Understood, it is not the catch all but, what if I am designing a server that needs to accept calls from 15 or more 1800 numbers? How would you now channel it to a catch all? Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Sunday, June 03, 2007 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] really strange behavior In short, the 's' extension is not a catch-all. The use of 's' can be confusing. The best example I have of the use of 's' is when a ZAP call comes in on an analog line. IIRC, the book says something to the effect that 's' is for when, upon arrival in a context, the call has no other place to go. Works for 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 ___ --Bandwidth 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] Loud noise instead of MOH
- Original Message - From: Mauro Zanin [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, June 03, 2007 12:32 PM Subject: [asterisk-users] Loud noise instead of MOH Hi Everybody, I'm experiencing this kind of issue. One ASTERISK 1.2.17 is connected to a Bristuffed HFC single ISDN channel card. Everything seems to work but sometimes the third party caller when listening to MOH listens some SSH! instead of MOH, this is not continuos, MOH plays ok for, say, 20 seconds then the sound and then another 30 seconds of good MOH. We have some SIP phones for extensions (Grandstream 496 ATA and One SPX2000). Sometimes also ATAs hung, not really got ATAs at all. Does Anybody have some hint? Ciao a tutti Mauro Use native format .wav to play your moh instead of .mp3 may solve your problem. ___ --Bandwidth 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] SIP Options Reply Ignored
Hi I have FC6 system in the office running SVN-trunk-r63567 It is behind a NAT router which I have configured to do port forwarding etc. Asterisk connects and registers correctly to my SIP service (Sipgate.co.uk) and I can make and receive calls from any SIP phone on the office LAN. The problem comes when I try to use a SIP phone at home (also behind a NAT router). The phone registers correctly and I can see the SIP OPTONS packets being sent to the phone (SNOM 190). I can see an OK reply being received by Asterisk (using SIP DEBUG). However the OK reply appears to be ignored and a retransmission is made and the phone is marked as UNREACHABLE and will not accept any calls. Any ideas? Ian C ___ --Bandwidth 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] Strange problem with channel allocation
Hello I just settup a realtime mysql table for sip_peers. All peers
(friends) is autenticateing but when i want to initiate a call between them
i got the following error. Someone have some ideea? Thank you.
---Cut Here---
pbx*CLIconsole dial 1014
== Console is full duplex
-- Executing [EMAIL PROTECTED]:1] Dial(OSS/dsp, SIP/1014|40|t) in new
stack
[2007-06-03 20:16:10] DEBUG[27424]: res_config_mysql.c:650 mysql_reconnect:
MySQL RealTime: Everything is fine.
[2007-06-03 20:16:10] DEBUG[27424]: res_config_mysql.c:138 realtime_mysql:
MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '1014'
-- Called 1014
[2007-06-03 20:16:10] WARNING[27424]: channel.c:3222
ast_channel_make_compatible: No path to translate from
SIP/1014-081e93c0(256) to OSS/dsp(64)
[2007-06-03 20:16:10] WARNING[27424]: channel.c:3222
ast_channel_make_compatible: No path to translate from
SIP/1014-081e93c0(256) to OSS/dsp(64)
^ ??
[2007-06-03 20:16:18] NOTICE[27408]: chan_sip.c:2758 auto_congest:
Auto-congesting SIP/1014-081e93c0
[2007-06-03 20:16:18] NOTICE[27408]: chan_sip.c:2758 auto_congest:
Auto-congesting SIP/1014-081e93c0
-- SIP/1014-081e93c0 is circuit-busy
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:210 mysql_log:
cdr_mysql: inserting a CDR record.
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:226 mysql_log:
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield)
VALUES ('2007-06-03 20:16:10','','','s','default',
'SIP/1014-081e93c0','','','',8,0,'NO ANSWER',3,'','')
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] VoiceMail(OSS/dsp, u1014) in new stack
Console call has been answered
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6798 vm_exec:
Prefixing the mailbox with an option is deprecated ('u1014').
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6798 vm_exec:
Prefixing the mailbox with an option is deprecated ('u1014').
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6799 vm_exec: Please
move all leading options to the second argument.
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:6799 vm_exec: Please
move all leading options to the second argument.
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:2854 leave_voicemail:
No entry in voicemail config file for '1014'
[2007-06-03 20:16:18] WARNING[27424]: app_voicemail.c:2854 leave_voicemail:
No entry in voicemail config file for '1014'
-- Executing [EMAIL PROTECTED]:3] Hangup(OSS/dsp, ) in new stack
== Spawn extension (default, 1014, 3) exited non-zero on 'OSS/dsp'
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:210 mysql_log:
cdr_mysql: inserting a CDR record.
[2007-06-03 20:16:18] DEBUG[27424]: cdr_addon_mysql.c:226 mysql_log:
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield)
VALUES ('2007-06-03 20:16:10','','','1014','default',
'OSS/dsp','SIP/1014-081e93c0','Hangup','',8,0,'ANSWERED',3,'','')
Hangup on console
[2007-06-03 20:16:18] DEBUG[27370]: res_config_mysql.c:650 mysql_reconnect:
MySQL RealTime: Everything is fine.
[2007-06-03 20:16:18] DEBUG[27370]: res_config_mysql.c:138 realtime_mysql:
MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '1014'
---And Here---
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Re: [asterisk-users] SIP Options Reply Ignored
On Sun, 3 Jun 2007, Ian Clough wrote: The problem comes when I try to use a SIP phone at home (also behind a NAT router). The phone registers correctly and I can see the SIP OPTONS packets being sent to the phone (SNOM 190). I can see an OK reply being received by Asterisk (using SIP DEBUG). However the OK reply appears to be ignored and a retransmission is made and the phone is marked as UNREACHABLE and will not accept any calls. Wait, so this is the phone registering to Asterisk? Any inconsistencies in the source/destination ports vis-a-vis the NAT state pinholes? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth 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] wifi sip phone real-world experiences?
I've tested a few different wifi SIP phones for office/factory use, and generally have been underwhelmed. Before I grab another few and test, I'd like to ask around here about the candidates. My requirements are relatively simple: - WEP/PSK should be supported WITHOUT dragging the phone down - roaming between access points without dropping the call - decent set of ringers, not the garbage that cell phones use now - vibrate, ring, ringvibe (increasing volume a bonus) - transfer, hold, display caller ID - mass deployment (similar to polycom?) TFTP/FTP/HTTP config - decent, but not enormous battery life - replaceable batteries What's not important: - NAT passthrough - colour screen - MIDI ring tones Bonus features: - programmable soft buttons - bluetooth Currently I'm looking at the WIP300/330 and if I can find a source, the CW/Hitachi wifi phone. I've tried some of the UTStarCom phones, Pulver's WiSIP and another I can't think off offhand. Is there anything out there that anyone else has been reasonably happy with? -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] asterisk auto dial does not wait for answer
Stephen Bosch wrote:
[outbound-reminder]
exten = _01N.,1,Dial(Zap/g1/${EXTEN},20)
Zaptel considers the call answered the moment it is bridged. You need
call progress detection of some kind. Are you using an analog card?
This is not fully correct.
Calls going out FXO signaled ports are considered answered as soon as
dialing is finished. This applies to analog FXO ports, as well as FXO
signaled T-1 channels. It is not specific to Asterisk or Zaptel.
Sangoma cards have the same issue. ATAs with FXO ports have the same
issue, so do channel banks, as do most PBXs.
This issue does not apply to FXS, EM, PRI, VoIP, or most any other port
types.
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Re: [asterisk-users] really strange behavior
To catch 1 or more of any character you would use _. To catch 2 or more digits you would use _X. You almost NEVER EVER EVER want to use _. as it will also match extensions like s, a, o, and h. Tom Lynn wrote: exten = _X.,1, On 6/3/07, BSumrall [EMAIL PROTECTED] wrote: Understood, it is not the catch all but, what if I am designing a server that needs to accept calls from 15 or more 1800 numbers? How would you now channel it to a catch all? ___ --Bandwidth 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] really strange behavior
BSumrall wrote: Understood, it is not the catch all but, what if I am designing a server that needs to accept calls from 15 or more 1800 numbers? You put all 15 or more 800 numbers in as specific individual extensions just like most every one else does. If your numbers have a pattern like 800-512-8000 - 800-512-8099 then you could use _80051280[0-9][0-9] ___ --Bandwidth 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] wifi sip phone real-world experiences?
Did you look at this one? No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 We're thinking of buying a couple for communication between our IT team members across 5 floors in 2 buildings. If you've tried it I'd be interested to hear you review. Cheers Alez ___ --Bandwidth 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] wifi sip phone real-world experiences?
On Sun, 3 Jun 2007, Alex Crow wrote: Did you look at this one? No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a UT StarCom F1000G ... I have one of those phones and it's not very good - certianly can't handle roaming between APs on the same networks at all. It's still firmly in the toy category for me, I've *never* subject my paying clients to it! We're thinking of buying a couple for communication between our IT team members across 5 floors in 2 buildings. Forget it and get a SIP based DECT phone with proper DECT repeaters. Look at the Siemens range. I'm using the C460IP's and range coverage with a single wall mounted base unit is 10x better than WiFi. Repeaters are avalable to extend the range too, and it all just works. Or just get a decent contract on your mobile phones and use them with a GSM base unit with a SIM on the same contract. (to enable desk phone to mobile communcation at the contract rate and vice versa) I've used this: http://www.westlake.co.uk/Cellroute_GPRS_GSM_Fixed_Cellular_Terminal_gateway.htm with good results, although it's analogue, it does the job and is cheap enough for most small systems. 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] zaptel on CENTOS servercd
Khaled Chehab wrote: I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there is someone did that, Please in need to have the installation procedure step by step. Its too urgent for me . Thanks alot These scripts are working pretty well for me. They are based on a script I found on the wiki. # Script to download pre reqs for installation # == #!/bin/bash rpm --import http://mirror.centos.org/centos/RPM-GPG-KEY-CentOS-4 yum -y install kernel-source bison openssl-devel yum -y update # Use kernel-devel instead of kernel-smp-devel # if not a SMP machine # |==| | | yum -y install gcc kernel kernel-smp-devel bison openssl-devel shutdown -r now # end script # Script to download asterisk, etc and build/install # == #!/bin/bash rm -f /usr/lib/asterisk/modules/* cd /usr/src rm -rf asterisk wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.18.tar.gz tar -zxvf asterisk-1.2.18.tar.gz mv asterisk-1.2.18 asterisk cd /usr/src rm -rf zaptel wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.17.1.tar.gz tar -zxvf zaptel-1.2.17.1.tar.gz mv zaptel-1.2.17.1 zaptel cd /usr/src rm -rf asterisk-addons wget http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.2.6.tar.gz tar -zxvf asterisk-addons-1.2.6.tar.gz mv asterisk-addons-1.2.6 asterisk-addons cd /usr/src rm -rf asterisk-sounds wget http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1.tar.gz tar -zxvf asterisk-sounds-1.2.1.tar.gz mv asterisk-sounds-1.2.1 asterisk-sounds cd /usr/src/zaptel make clean make install make config cd /usr/src/asterisk make clean make install make samples make config cd /usr/src/asterisk-addons make clean make install cd /usr/src/asterisk-sounds make install -- 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] wifi sip phone real-world experiences?
On Sunday 03 June 2007 4:30 pm, Alex Crow wrote: No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a terrible phone. I've tried them. the screen is pretty much useless, the buttons are *TINY*, the battery life horrible, and the ringtones gimmicky. I haven't tried WEP or WPA on these things, but the phones I've gotten rid of long ago due to their problems. -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] Auto Dial Problem
Steve Totaro wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Saturday, June 02, 2007 11:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto Dial Problem
[EMAIL PROTECTED] wrote:
Hi All,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
My sample call file :
hannel: local/[EMAIL PROTECTED]
MaxRetries: 5
RetryTime: 300
WaitTime: 40
Account: Reminder
context: remindem
extension: s
priority: 1
Set: MSG=0135.20070601.0124787924
Set: APPTDT=20070601
Set: APPTTIME=0135
Set: APPTPHONE=0124787924
Set: CALLATTEMPTS=5
Set: CALLDELAY=300
My outbound-reminder context:
[outbound-reminder]
exten = _01N.,1,Dial(Zap/g1/${EXTEN},20)
My remindem context :
[remindem]
exten = s,1,Answer()
exten = s,2,Wait(2)
exten = s,3,Playback(custom/reminder5)
Once asterisk start to execute .call file, my handset
rings but the console shows Playback(custom/reminder5)
I believe that it is because you are using zap lines to dialout. Zap
lines are considered answered almost immediately. The believe digital
and VoIP channels on the other hand have the call supervision that can
distinguish when an answer is made.
Any kind of dialout like that, I just use my sip service provider.
--
Warm Regards,
Lee
This may be true with analog zap channels but not T1 PRIs.
Additionally, some VoIP providers answer the call prior to initiating
the second leg of the call.
Thanks, I wasn't aware of that. I'm still getting my feet wet with 4-10
extension installs.
Who is your provider that does not give you an answer until the call is
really answered?
www.axvoice.com
-- Executing Macro(SIP/111-08e74378,
DialOutside|SIP/axVoice/302381) in new stack
-- Executing GotoIf(SIP/111-08e74378, 1?2:4) in new stack
-- Goto (macro-DialOutside,s,2)
-- Executing Dial(SIP/111-08e74378, SIP/axVoice/302381||T)
in new stack
-- Called axVoice/302381
-- SIP/axVoice-08e798b8 is making progress passing it to
SIP/111-08e74378
-- SIP/axVoice-08e798b8 answered SIP/111-08e74378
I had to test it again to be sure. The last output line indicating the
channel was answered was outputted by the CLI only after I answered my
cell phone.
Last I checked, IAX.cc (now Vitelity) was giving me answered
immediately. I am not sure that is the case anymore.
I've only had axvoice and telasip. Can't remember if telasip worked the
same way or not unfortunately.
--
Warm Regards,
Lee
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[asterisk-users] Can two card be configured on same machine.
I have a 2 sangoma cards that need to be configured on a same server, one is a T1 and another is a for PSTN line. Is this possible, if so please help. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] linksys wip300 firmware
Firefox under Linux seems to have loaded it in ok. No browser under OSX worked (PPC or Intel), including Camino, Firefox, or Safari. On 6/3/07, Bryan Laird [EMAIL PROTECTED] wrote: If your getting an error about invalid Octet key or something make sure you are doing the load from a Windows Computer it's been noted that a lot of people have had problems loading that version from various OS's. I went through a few computers before it took the load (for the record it worked from winxp / ie7) On Jun 2, 2007, at 7:48 PM, Ilan Rabinovitch wrote: Hello, Does anyone have access to version 1.00.07 of the Linksys WIP 300 firmware? The only version on their site at the moment is 1.00.09, which the phone refuses to load. Regards, Ilan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth 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] Can two card be configured on same machine.
Sangoma offers EXCELLENT Technical support Why not try them first? They have never failed me yet, even for our peculiar requirements Email address on their website John Novack Sanjay Rajdev wrote: I have a 2 sangoma cards that need to be configured on a same server, one is a T1 and another is a for PSTN line. Is this possible, if so please help. Regards, Sanjay Rajdev ___ --Bandwidth 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] click to call
Hi Gordon So, mexuar solution was that java softphone that you talked about? Any other small softphone type solution around, something on the same lines of what you described, something that the user could download but could be preconfigured or passed parameters to so they user wont have to mess with settings. Regards AK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Sábado, 02 de Junio de 2007 03:09 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] click to call On Fri, 1 Jun 2007, Anton Krall wrote: So Guys, no go on this topic? I trialled a click-to-dial application recently. It generated a lot of controversy on the list (search the archives) because various people said it couldn't be done/wouldn't work, etc. Then there were whinges about the commercial nature of the application (it's licensed, not free, and details were being posted to the -users list) and so on. Personally, I didn't see why as the creators of the code were simply replying to questions asked by list members, however... (That's probably why you've not gotten many replies ;-) So the thing I trialled was a button on a web page which downlaoded a soft-phone program written in Java to your browser. The soft-phone uses the IAX protocol to connect to an asterisk server, then depending on the javascript that you write to encapsulate the button on the web page, you have the ability to specify username password (to authenticate back to the asterisk server) and number to dial - the number you dial could even be entered via more javascript on the webpage, and the asterisk server at the back-end can then do what it needs to do with the number - dial an extension in a closed system, or even initiate a dial-out to the PSTN, if the server as such a connection and the connection is authorised. The end-user pushing the button doesn't need to see any of this at all - it can all be embedded in the javascript behind the button. You can specify callerId too, or dial different numbers, so the person answering the call could use this information to know what web page you are on for example. You can even embed it into an email signature with a different number then you could tell if they are calling you in reply to an email, and so on. (And much as I hate big HTML based email signatures, if done correctly this could be quite effective - and it doesn't need to download the Java - about 120KB until you click on the button) (They have a demonstration client which works with the Tesco VoIP service - you enter your Tesco username/password, then get a phone application with buttons, etc. The Tesco VoIP system unusually uses IAX rather than SIP as their transport mechanism!) I tried the application on a WinXP box, Linux box and Mac, and as long as the sound system was setup to work with the headset microphone, it just worked - At last, Java doing what it was supposed to be doing, working correctly cross platform! Some of the whinges to the list were that a soft-phone couldn't possibly be written in Java as Java was too heavyweight - well, this is the latter part of the first decade of the new millennium and Java has come a long way since it was first released, and they couldn't be further from the truth - in use on my 2GHz Linux box, it was using about 2-3% CPU, and at 120KB to download, is no worse than your average mid-resolution camera image these days. If this is what you're after, then go to http://www.mexuar.com/products_connect.shtml They were happy to give me a time-limited trial of the software, which I used, and found worked really well. You will need to write some html and javascript to encapsulate it into your own web page, but that's not hard to do and examples are provided. Now all I need is some clients to sell it to ;-) Gordon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Jueves, 31 de Mayo de 2007 10:58 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] click to call The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples or
RE: [asterisk-users] click to call
True, maybe I didnt make myself clear on that point, what i meant was, Im not looking for an app that would let people click a sip: URL type to make a call using their already installed softphone but rather allow any user that visits our website to click on something and either open a web softphone or download a small one thats preconfigured and allow them to call a predefined extension on our asterisk server. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sábado, 02 de Junio de 2007 08:23 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] click to call So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I think the misconception is on your part. No matter what, the client will have to run some sort of softphone application. Whether it is implemented in Java, an exe, ActiveX, or some other 3rd party app. There is no magic image that makes phone calls. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Friday, June 01, 2007 10:53 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] click to call So Guys, no go on this topic? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Jueves, 31 de Mayo de 2007 10:58 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] click to call The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? What we have done in the past is created url's like this : sip:4044565941. Xlite will register itself as the sip handler on your system. If you want a generic click to call (ability to call numbers on any given website) check out moziax - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Anton Krall.vcf Description: Binary data ___ --Bandwidth 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] click to call
Thank you for the explanation Dean, you are right on the money and could be more precise. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Sábado, 02 de Junio de 2007 04:34 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] click to call Joseph, This issue is people get confused; Click-to-Call - the ability to enter a number on a web page (or 2 numbers in the case of apps like JaJah) and have a centralized server deliver a two legged outbound call resulting in a 2 (or more) party conference call. This is inbuilt in Asterisk and can be found/implemented very easily by referring to the voip-info site for Dynamically generated call files. Click-to-Talk is different. It is the ability to using a browser to visit a web site, by clicking on a hyperlinked image or initiating the call in some other way your browser downloads either a java applet in the case of Mexuar, JiaxClient, Barbizan and a few other java solutions or an Active-X client in the case of Estara. These applets are basically installing a softphone onto the browser that is configured to dial a particular extension eg throught to sales or technical support. Its frustrating people confusing the terms but hopefully over time people will understand the differences. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joseph Bajin Sent: Saturday, 2 June 2007 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call You shouldn't need a softphone to do Click to Call.. The idea is pretty simple, and maybe I am missing something since I am haven't worked with Asterisk enough, but basically you start off by making the call to the Initial Party, Park the Call, Call the Other Party and then Connect them together.. Seems pretty simple and easy enough to do. ___ --Bandwidth 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_mobile issue
On 6/3/07, Steve Totaro [EMAIL PROTECTED] wrote: Hello, I just did a fresh svn install of 1.4 trunk everything. Everything compiles and installs just fine. When I get to asterisk-addons, I cannot select chan_mobile in menuselect. Chan_mobile is not even an option in menuselect for asterisk trunk. I tried the latest patch which failed in many places but did add an option for chan_mobile in menuselect for asterisk but it still cannot be enabled. If chan_mobile is in the asterisk-addons trunk then I must just be missing something on where to enable it, right? The readme says nothing. This box is fedora core 6 with all the bluez stuff installed and loaded and a dongle attached. I can see and pair with the box with my cell phone so Bluetooth is working in linux. Ideas? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB Do you have the devel package for libbluetooth2 (or libbluetooth) installed? chan_mobile won't be an option without that package installed. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
