Re: [Asterisk-Users] How to send Fax from Asterisk
On Thu, 2005-07-21 at 13:00 +0900, Vic wrote: Dear all, 1. Person will call our phone number 2. He will be asked to press 1 for Office 1 map, 2 for Office 2 map and 3 for Office 3 map. 3. User presses 1. 4. User is asked to enter his phone number. 5. User enters his phone number and hangs up. 6. Asterisk calls the number entered by user and sends a fax. Can it be done? Tom Rymes [EMAIL PROTECTED] wrote: If you install Hylafax, I'm sure that you could have the person choose which map they want and then enter their fax number. You can then call sendfax from extensions.conf with the relevant data to send the fax. Hylafax would then send the fax out using its own dedicated analog or t1/e1/whatever modem (ie: not through asterisk b/c that seems to be unreliable at best). The wiki has information on originating calls from agi scripts. Bill ___ Sorry for the bad quoting but the original has a strange format. IMHO the above would answer your needs. Hylafax is a specialist in faxing, it just works. All you need is a bit of glue, the hooks are already in place. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play Dialtone - get digits
On Wed, 20 Jul 2005, Ed Greenberg wrote: I'd like to write a snippet of dialtone that plays dialtone and collects a specific number of digits into a variable. Sort of like READ but with a generated dialtone. Naturally, I want the dialtone to stop playing after the first digit. I can't find this anywhere. Only thing I can think of is a no-password DISA. Is this the correct method? Is there a better one? DISA would proably work, though it may be a hassle since the call will be sent into the disa context. Another option is to use READ with a filecontaining a recording of the dialtone. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is soekris good?
On Thu, Jul 21, 2005 at 09:59:05AM +0800, Ronald_Wiplinger wrote: asterisk_on_oelf wrote: Hi, I have a soekris 4801 since some days. I use it with a FritzCard-USB and an internal HFC-Card (NT Mode). Everything is working, but I still havn't had time for performance test. Only thing I tested, was two ISDN channels via FritzCard in a conference room. CPU usage was nearly 70% I hope next weekend I'll find more time. What WiFi phone do you want to use? I tried a ZyXEL P2000W, but voice quality was very bad. Jens, I am trying to find out what is the best board for us. I want to build an asterisk based PBX with one digium TDM422 card. A USB wireless adapter should make the entire system with: * [EMAIL PROTECTED] (Is @home or regular better?) [EMAIL PROTECTED] approach is to give you a toolbox for building Asterisk on your system. This works fine for a PC, but generates an over-inflated system if you want to generate an embedded-type system. Now from my point of view: what are your recommendations for building a CF-based system from based on Debian Sarge? * Shorwall firewall Why shorewall? Why not firewall? I believe that the rules shorewall creates are not very efficient, latency-wise. * QoS * Hotspot for wireless phones * web server for Asterisk (billing, settings) - maybe thttpd since it also can IPv6 * IPv6 in the second step (I think @home cannot IPv6) * astcc * h.323 module * wakeup * festival (Maybe the CPU / RAM is too low for that) * MOH * voice mail * ??? What do you think about it? Is the 4801 right for that? Some of that is CPU intensive. Festival, MOH(?), astcc(?), voip with a compressed codec. The sokeris box may be enough for a light load, though. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h: not found
On Thu, Jul 21, 2005 at 01:31:59AM +0800, chris wrote: hi kevin, i tried removing the enitre asterisk directory and upadatesd my cvs folder. and try to run make.. i'm getting make_version_h : cannot execute error Maybe you misread the error? Maybe this is an error from this script that it cannot execute something else? Can you execute it manually? If so, add '-x' to its first line to get traced execution. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GXP2000 resetting all the time
I have all my GXP-2000's set to dynamic with no problems. You need to make sure they have the latest firmware, as this fixed a few issues and improves the overall usage of the phone. Hopefully they will make the useless LED's work so we can line monitor etc... Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan C. Smith Sent: 20 July 2005 18:46 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Grandstream GXP2000 resetting all the time Do you have the address set to dynamic or static in sip.conf? -Original Message- From: Michiel van Baak [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 20, 2005 1:59 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Grandstream GXP2000 resetting all the time On 16:06, Wed 20 Jul 05, [EMAIL PROTECTED] wrote: All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly appreciated. Marc Hi, Are you using the latest firmware on the phones ? We use 1.0.1.9 and have no problems at all. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/53 - Release Date: 20/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/53 - Release Date: 20/07/2005 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] That is not a valid conference number.. with ztdummy running
On Wed, Jul 20, 2005 at 01:41:52PM -0400, O'Neill,Davin S. wrote: I previously had Asterisk 1.0.7 running on a Linux 2.4.x kernel with ztdummy. I was able to do things like meetme and music on hold. I recently installed Asterisk 1.0.9 on a different machine with a Linux 2.6.x kernel running ztdummy. I installed and configured everything the same way, but when I try to call into a conference room I get the error message stating, that is not a valid conference number My first guess: udev. README.udev . If that is not good enough, please try to demostrate (to yourself and to us) that you have invastigated enough. Why do you think that that error message is relevant to ztdummy/zaptel? Can you back this up with more specific error messages? And can you use zaptel timing? try running zttest, if possible, as the same user that runs asterisk. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns quadBRI on Dell PowerEdge
David Hajek wrote: Yes, I tried signalling = bri_cpe_ptmp. When I put the card into older system and use same cables, same ISDN units, same Asterisk configs (but older bristuff!) it works fine. When I put the card into Dell, I got the CRC errors as I wrote before. Maybe someone from Junghanns is watching this thread and can give some help? Can you try the 'older' bristuff on that DELL ? That seems to be the difference between your 2 systems.. You can also try swapping the cable (as in the other post) and try a different PCI slot in your Dell. Cheers Kristof. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOWTO capture digits
On Wed, Jul 20, 2005 at 02:13:57PM -0400, J.Raborg wrote:
Folks:
does anybody have an idea? how to capture the DTMF digits to a file, after
an extn asnwer? then POST it to a url?
Off the top of my head:
Read(DIGITSVAR)
System(echo ${DIGITSVAR} /path/to/file)
Curl(URL)
;or:
;System(wget --with --the -right --switches ${DIGITSVAR} URL)
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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Re: [Asterisk-Users] Problem while configuring two TDM400P cards
Hi to all once again Thanks for you help. I always get my problem solved from here. What i did. export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout -r v1-0 zaptel libpri asterisk and then compiled each of them then on running /etc/init.d/zaptel start Both of cards are configured now and their LED's are on now. But the problem is that I cannot get asterisk work with my old files. I am getting the following errors in my Asterisk-messages log file Jul 21 11:55:10 WARNING[1607]: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: pbx_substitute_variables_varshead Jul 21 11:55:10 WARNING[1607]: Loading module pbx_dundi.so failed! Regards, Mazhar On 7/20/05, Watkins, Bradley [EMAIL PROTECTED] wrote: What revision of card is the new one? It sounds like you have one of the new Rev I cards and you aren't running either 1.0.9 or CVS HEAD. Either of these will solve your problem if I am correct. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mazhar Hussain Sent: Wednesday, July 20, 2005 7:22 AM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem while configuring two TDM400P cards Hi to all, I have been using Wildcard TDM400P with four fxs modules and it was working fine now I have added another Wildcard TDM400P with four fxs modules . So there are total 8 ports for 8 hard phones. I have modified following configurations In /etc/zaptel.conf loadzone=us defaultzone=us fxoks=1-8 in /etc/asterisk/zapata.conf context=headoffice signalling = fxo_ks callerid=HeadOffice channel = 1-4 ;channel = 1-8 /sbin/modprobe wcfxs works fine ans show eight modules mapped. But when I run /sbin/ztcfg -vvv I get following error ZT_CHANCONFIG failed on channel 5 Is there error in configuration I also have checked Diguim card separately it works This I found in /var/log/asterisk/messages Jul 20 11:57:04 WARNING[1144]: Unable to specify channel 5: Device or resource busy Jul 20 11:57:04 ERROR[1144]: Unable to open channel 5: Device or resource busy here = 0, tmp-channel = 5, channel = 5 Jul 20 11:57:04 ERROR[1144]: Unable to register channel '1-8' Jul 20 11:57:04 WARNING[1144]: chan_zap.so: load_module failed, returning -1 Jul 20 11:57:04 WARNING[1144]: Loading module chan_zap.so failed! I will be very tanksfull to for this help. As I always get my query solved from here. Regards, Mazhar Nettechltd.com The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is soekris good?
Ronald_Wiplinger wrote: asterisk_on_oelf wrote: Hi, I have a soekris 4801 since some days. I use it with a FritzCard-USB and an internal HFC-Card (NT Mode). Everything is working, but I still havn't had time for performance test. Only thing I tested, was two ISDN channels via FritzCard in a conference room. CPU usage was nearly 70% I hope next weekend I'll find more time. What WiFi phone do you want to use? I tried a ZyXEL P2000W, but voice quality was very bad. Jens, I am trying to find out what is the best board for us. I want to build an asterisk based PBX with one digium TDM422 card. A USB wireless adapter should make the entire system with: * [EMAIL PROTECTED] (Is @home or regular better?) * Shorwall firewall * QoS * Hotspot for wireless phones * web server for Asterisk (billing, settings) - maybe thttpd since it also can IPv6 * IPv6 in the second step (I think @home cannot IPv6) * astcc * h.323 module * wakeup * festival (Maybe the CPU / RAM is too low for that) * MOH * voice mail * ??? What do you think about it? Is the 4801 right for that? Some questions about Soekris: What is in the package? (Power adapter?, CF?, manual? ...) How to install it? What is the CF size you are using? and how much is still free? What have you installed? We are in the process to develop a WiFi phone, bye Ronald Everyone, [EMAIL PROTECTED] is a distribution that includes Asterisk. Asterisk is an application. [EMAIL PROTECTED] cannot be compared to Asterisk, they are not the same thing. With that being said, [EMAIL PROTECTED] is basically CentOS that installs Asterisk by default, Myself and others have pointed this out several times on this list. Secondly, you would NEVER want to run [EMAIL PROTECTED] on a Soekris or any embedded device. It's WAY too huge, plus running it from flash would be a bad idea. Third, the Soekris Net4801 will probably not be able to handle the TDM422. Echo cancellation and transcoding will probably bring it to it's knees. As far as IPv6, Asterisk cannot do IPv6 anyways (I know there were some patches at one time, but I don't think they are current). Use a Mini-PCI wireless card instead. Overall, it looks like you are trying to cram WAY TOO MUCH functionality into one box, especially something like the Soekris Net4801! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF with Asterisk as SIP client
Hello, I have the following setup: sip phones -SER - asterisk - voip provider1 - voip provider2 i got a toll-free DID from voipprovider1 to allow people from outside to call into asterisk, get authenticated, and use voipprovider2 to call out (kind of a primitive calling card app). anyway, voiprovider is giving my bad DTMF (usually 2 keypresses for every 1)...transport is via SIP, i am registered in sip.conf with a register statement (i.e. asterisk is a SIP client) and ulaw and alaw are the first allowed codecs. When i set dtmf as info or RFC2833 i don't get any tones, and when i set inband i'm back to bad DTMF. if i call into the extension from one of my sip phones (i.e. not via voip provider) and interact with the menu (put in my authentication and dial the onward number) it works fine. anyone come across this? any tips on how to solve it? any help is appreciated, thanks, yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h: not found
hi Tzafrir, i was able to run make by removing ^M at the end of each line of each script, i also checked all script file on the /asterisk folder and execute dos2unix command on all script files, however when i run make i encountered another problem. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/in clude -D_REENTRANT -D_GNU_SOURCE -O6 -Wcast-align -DSOLARIS -DBUSYD ETECT_MARTIN -fomit-frame-pointer-c -o md5.o md5.c md5.c: In function `byteReverse': md5.c:47: warning: cast increases required alignment of target type md5.c: In function `MD5Update': md5.c:98: warning: cast increases required alignment of target type md5.c:107: warning: cast increases required alignment of target type md5.c: In function `MD5Final': md5.c:142: warning: cast increases required alignment of target type md5.c:153: warning: cast increases required alignment of target type md5.c:154: warning: cast increases required alignment of target type md5.c:156: warning: cast increases required alignment of target type gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/in clude -D_REENTRANT -D_GNU_SOURCE -O6 -Wcast-align -DSOLARIS -DBUSYD ETECT_MARTIN -fomit-frame-pointer-c -o term.o term.c In file included from include/asterisk/utils.h:26, from term.c:32: include/asterisk/strings.h:232: parse error before `va_list' include/asterisk/strings.h:232: warning: function declaration isn't a prototype make: *** [term.o] Error 1 bash-2.05# any ideas on how i can fix this? thnks in advance. chris. - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, July 21, 2005 2:51 PM Subject: Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h: not found On Thu, Jul 21, 2005 at 01:31:59AM +0800, chris wrote: hi kevin, i tried removing the enitre asterisk directory and upadatesd my cvs folder. and try to run make.. i'm getting make_version_h : cannot execute error Maybe you misread the error? Maybe this is an error from this script that it cannot execute something else? Can you execute it manually? If so, add '-x' to its first line to get traced execution. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
I think that's mostly right, but it should also be a native xfer function working the same way regarding of the user agent, some sort of common ground we can trust for installation with mixed devices. By the way: anyone got experience in attended trasfer with snom ? :) Alessio Focardi PF Oh, you mean the completely natural feeling put them on hold, dial PF new party, tell them you have a transfer, hit transfer? I want some of PF whatever kool-aid the person who thought that one up had. I still feel PF like I'm losing a call every time I do an attended transfer. In my opinion there should be only one transfer function, let suppose it's called by #. - You get a call - You want to transfer it - You hit # - You are presented a tone - You dial the extension you want to transfer to Now the hard part - If you hang up prior of the other party has answered you get an unattended transfer if, for any reason the other party dont answer (busy, no answer, wrong extension etc) call should be bounced back to you - If you stay on the phone and the other party answers you talk to him, introduce the call then hitting # again will switch back and forth between the call you are tranfering and the transfer party if you hang up call is trasfered to the other party if the other party hangs up you get back to the original call Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. What do you think about this flow ? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failover question
On Thu, Jun 30, 2005 at 09:45:56AM -0600, Joseph wrote: I think this is a weak point in asterisk. It doesn't even have a means of email notification if IAX or SIP registration fails. But it sends an error to the a log (configurable to some extent in logger.conf). tail -f /that/log/file | grep your_favourate_filter | \ while read message do parse that message mail to admin done This is a simple and primitive shell script. There are quite a few nice log watchers out there to do that. This would need to be added to the list of priorities. But I'm not sure who to address to. Write a patch or get someone to write that patch for you. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP messengers video phones
Is there a possibility to send text based messages from/to a sip phone (text display) or to a video phone or from/to a messenger? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is soekris good?
I am trying to find out what is the best board for us. I want to build an asterisk based PBX with one digium TDM422 card. A USB wireless adapter should make the entire system with: * Asterisk at home (Is @home or regular better?) I have no experiences with @home. I use a debian. Debian sarge includes asterisk 1.0.7. Debian unstable includes 1.0.9. * Shorwall firewall * QoS I don't know shorwall. I use iptables with stateful inspection and HFSC shaping as described in the wiki pages. * Hotspot for wireless phones * web server for Asterisk (billing, settings) - maybe thttpd since it also can IPv6 * IPv6 in the second step (I think @home cannot IPv6) * astcc * h.323 module * wakeup * festival (Maybe the CPU / RAM is too low for that) * MOH * voice mail * ??? What do you think about it? Is the 4801 right for that? I'm afraid CPU and RAM is to low for all that things (some test result will follow in the next days) The TDM422 could be to large for the case provided by soekris, so you have to use your own case. Some questions about Soekris: What is in the package? (Power adapter?, CF?, manual? ...) In the package is only the board (in the case if orderd with case). You have to order power adapter too. The manual you can found on the web page. CF is not included. How to install it? http://www.soekris.com/support.htm I used Running Debian Linux, by Mike Machado After the initial installation, I updated to standard debian via Internet. What is the CF size you are using? and how much is still free? What have you installed? First I used a 256MB CF card. It is enought for the installation described in my last mail. Round about 60MB was still free. But now I added hylafax. That need a lot of space, because it needs ghostscript to convert and ghostscript wanted some x-libraries. :-( So I have replaced the 256MB CF with a 2GB micro drive (only ~350MB used at the moment) . This has some more advantages. I think smart drive have more read/write-cycles. Thats better for voicemail and fax. bye Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel make problems (long)
I know that this subject has been treated in the past! As a matter of fact reading some old messages about compiling zaptel I made a couple of tests after the first compiling failure to understand why I can't compile on a specific machine, but I do not know how to handle the results. The machine has SUSE 9.3, and an updated kernel (2.6.11.4-21.7-default; as shown below). YAST (the graphical updater/installer/ect) tells me I have an installed kernel version 2.6.11.4-21.7-i586. It tells me furthermore that the kernel sources are in sync with the compiled kernel. Now I tried to compile zaptel both with the simple 'make' as well as with the make linux26. I get errors in the two cases. Simple make: [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 uname -r 2.6.11.4-21.7-default [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make clean rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o make -C /lib/modules/`uname -r`/build SUBDIRS=/home/aaberga/asterisk sources/zaptel-1.0.9 modules make[1]: Entering directory `/usr/src/linux-2.6.11.4-21.7-obj/i386/default' make -C ../../../linux-2.6.11.4-21.7 O=../linux-2.6.11.4-21.7-obj/i386/ default sources/zaptel-1.0.9 make[3]: *** No rule to make target `sources/zaptel-1.0.9'. Stop. make[2]: *** [sources/zaptel-1.0.9] Error 2 make[1]: *** [sources/zaptel-1.0.9] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.11.4-21.7-obj/i386/default' make: *** [linux26] Error 2 [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make linux26: [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 uname -r 2.6.11.4-21.7-default [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make clean rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core [EMAIL PROTECTED]:~/asterisk sources/zaptel-1.0.9 make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA - DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o make -C /lib/modules/`uname -r`/build SUBDIRS=/home/aaberga/asterisk sources/zaptel-1.0.9 modules make[1]: Entering directory `/usr/src/linux-2.6.11.4-21.7-obj/i386/default' make -C ../../../linux-2.6.11.4-21.7 O=../linux-2.6.11.4-21.7-obj/i386/ default sources/zaptel-1.0.9 make[3]: *** No rule to make target
Re: [Asterisk-Users] SIP messengers video phones
Ronald_Wiplinger wrote: Is there a possibility to send text based messages from/to a sip phone (text display) or to a video phone or from/to a messenger? Yes, there is in SIP if the SIP user agents support it. But no, Asterisk will not forward the SIP messages between the SIP user agents. Remember, Asterisk is not a SIP proxy. There's some work on creating a multiprotocol solution for instant messaging within Asterisk, but it will not be in the coming v1.2. /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Last two digits getting cut off?
Rob Engstrom [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) We've just setup our [EMAIL PROTECTED] server, with our quad port card. Everything works well so far. One thing I notice is that when I leave the handset on the hook and dial a #, all is well. If I pick up the phone and dial, it cuts off at 10 digits, which is a problem if I need to dial 1+area+phone # (12 digits). The phones are Poly Soundpoint IP 600's. I'm wondering if I've missed a config to allow more than 10 digits? The Cisco 7960 and Sipura devices have a dialplan that define when to send the dialled number to the server. If the SoundPoint has something similar, and I expect that it does, then that would be a good place to start looking. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 3 + AVM Fritz ?
On Thu, 2005-07-21 at 08:14 +1000, Eric Bishop wrote: Yes, I have some advice. Use Fedora Core 2. I have battaled for almost a year to get fcpci and udev-based distributions working with very limited success. On 7/21/05, Adrià Vidal [EMAIL PROTECTED] wrote: Someone have info about install an AVM fritz into FC3 ? I'm getting problems with kernelcapi, after succesfully installed the fcpci support. Thanks Too bad you had that experience. I have successfully used Asterisk (both stable and HEAD) on FC2, FC3, FC4 and Centos 4.1 without *any* issues. The reason why I would not go for FC2 is that Adria needs kernelcapi support. There have been many bugfixes in the capi modules in more recent kernels that are part of (updated) FC3, FC4, CentOS 4.1 and afaik not FC2. For that reason I would always use a recent FC distro like FC4. Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM gateway hardware
2 to 4 channels to start with. Allan. --- hakem voip [EMAIL PROTECTED] wrote: How many channels do you need per gateway ? I might have slution for you voip2gsm Regards On 7/20/05, Allan Kamau [EMAIL PROTECTED] wrote: Thanks Roger, I find the second option more interesting, let me know once you've managed to provide asterisk support for the GSM modem. Allan. --- Roger Schreiter [EMAIL PROTECTED] wrote: Allan Kamau schrieb: ... I am looking for a GSM VoIP gateway for use with Hi, do you think of something to interconnect to GSM carriers via cable (GSM-A) or do you think about using a GSM-modem with all its limitations? For the first option I could forward your email address to someone providing GSM-A stacks for asterisk. For the second option, it might be interesting for you, that we are currently also working on asterisk support for a GSM-modem. Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIDs in Thailand
Anyone know where to find a Thai DID to ring in SIP to asterisk? (probably Bangkok) Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thailand DIDs
Anyone know where to find a Thai DID to ring in SIP to asterisk? (probably Bangkok) Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and flash disks
On Wednesday 20 July 2005 15:49, Angus Comber wrote: Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and deleted immediately? I have been using flash-based asterisk and it works fine as long as you are careful about the maker - SanDisk did not work properly, we currently use the RiData flash disk and are happy with it. I have a Kingston Tech flash disk that was flaky and finally failed yesterday. They are not the fastest - running hdparm -tT on them reveals a speed of 2Mb/s which is about a third of the speed of 100Mbits ethernet. For call recording I usually add an IDE hard drive and make sure that most filesystems (e.g. /var,/tmp..) are loaded into a RAM disk Paul Hewlett -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thailand DIDs
Anyone know where to find a Thai DID to ring in SIP to asterisk? (probably Bangkok) Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mahler's Book - New Project
On Wed, 2005-07-20 at 10:39 -0700, Victor Rini wrote: David Stude wrote: #2, I'm planning to interface Asterisk with a Norstar MICS via PRI. Can anyone recommend a reference book or site more suited to this task? Sorry that link is kind of dead. I have the pdf if anyone is interested. Anything wrong with www.asteriskdocs.org ?? Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
Hello Adam, In my opinion there should be only one transfer function, let suppose it's called by #. AG Wrong, which other phone system have you used where every time you try AG and use some IVR that says Enter your xyz number followed by the # key AG and you end up being interrupted by asterisk to transfer the call ?? Well as you can see it was an example, actually you have to decide this mapping in features.conf, so what's the point ? Let say is *# or any other sequence :) Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. AG Nope, because if there are three parties: AG A - You AG B - Outside caller 1 AG C - Outside transfer party AG When you hangup, you don't want the other two legs to stay up, AG potentially forever depending on your hangup detection etc... I know what I want! :) Why not, I'm announcing a call, then going conference, then leaving because I already did my part, why the other 2 calls have to be disconnected ... because hangup detection works bad ? What do you think about this flow ? AG Any SIP phone (decent one) should have much more intuitive/instructive AG transfer process. All I'm asking is a native function that can be used regardless of the UA, if you got such functions integrated in the phone, better yet, is up to you to choose then. -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
On Thu, 2005-07-21 at 09:59 +0200, Alessio Focardi wrote: PF Oh, you mean the completely natural feeling put them on hold, dial PF new party, tell them you have a transfer, hit transfer? I want some of PF whatever kool-aid the person who thought that one up had. I still feel PF like I'm losing a call every time I do an attended transfer. In my opinion there should be only one transfer function, let suppose it's called by #. Wrong, which other phone system have you used where every time you try and use some IVR that says Enter your xyz number followed by the # key and you end up being interrupted by asterisk to transfer the call ?? Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. Nope, because if there are three parties: A - You B - Outside caller 1 C - Outside transfer party When you hangup, you don't want the other two legs to stay up, potentially forever depending on your hangup detection etc... What do you think about this flow ? Not only have you suggested pretty much what we have, except you've made it worse by taking away the # and * keys... If you are on a zap channel, just hook flash (or press flash/recall/whatever) and transfer the call, complete with conference option. Any SIP phone (decent one) should have much more intuitive/instructive transfer process. Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a ne pas voir
hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group Protek-on: CaraMail met en oeuvre un nouveau Concept de Sécurité Globale - www.caramail.com___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a ne pas voir
On Thu, 2005-07-21 at 10:41 +, ali kia wrote: hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group If that serves you better than this list or the existing irc channel (irc.freenode.net #asterisk) then by all means go for it, however I think you may find that getting a massive group to migrate to something new will be difficult. You may find that it is easier to use irc for real time chat and this list for email queries just because that is where everyone else already is. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with tdm11p
sometimes my tdm11p reads the caller id and sometimes doesnt read it and give me this : Jul 21 13:55:50 NOTICE[6284]: callerid.c:307 callerid_feed: Caller*ID failed checksum Jul 21 13:55:54 WARNING[6284]: chan_zap.c:5739 ss_thread: CallerID returned with error on channel 'Zap/4-1' what is the problem please help. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and flash disks
We had an experience with asterisk in a 512 MB IDE Compact Flash card. It uses nearly 400 MB of storage with a minimum installation of Linux, and 2.6.10 Kernel. I has web access with AMP Portal with needed modules as apache, php, etc. But the size can be less.It works fine. You can search about embedded asterisk. There are quite doc on the web. Regards. Sip ResearcherPaul Hewlett [EMAIL PROTECTED] wrote: On Wednesday 20 July 2005 15:49, Angus Comber wrote: Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and deleted immediately?I have been using flash-based asterisk and it works fine as long as you are careful about the maker - SanDisk did not work properly, we currently use the RiData flash disk and are happy with it. I have a Kingston Tech flash disk that was flaky and finally failed yesterday.They are not the fastest - running hdparm -tT on them reveals a speed of 2Mb/s which is about a third of the speed of 100Mbits ethernet. For call recording I usually add an IDE hard drive and make sure that most filesystems (e.g. /var,/tmp..) are loaded into a RAM diskPaul Hewlett-- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.zaTel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563-- ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a ne pas voir
On Thursday 21 July 2005 12:41, ali kia wrote: hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group also I would prefer not to switch to something M$ based... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems installing asterisk-addons
Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.capp_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 givenapp_addon_sql_mysql.c: In function `del_identifier':app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function)app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only onceapp_addon_sql_mysql.c:164: error: for each function it appears in.)make: *** [app_addon_sql_mysql.o] Error 1 I have set a password for root on mysql - could that be the problem? Should I remove the password? What is easiest way to do that? Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Bulletin Board for Asterisk is Now Available
without shattering what you are trying to do, asterisk wiki is the best effort in existance im my opinion. IRC and mailing list.. I dont think we need any more.. Keeping in mind im a nobody ;) On 19/07/05, matt001 [EMAIL PROTECTED] wrote: if it's of no use, we can always convert it for other type of forums. On Monday 18 July 2005 21:53, matt001 wrote: hi guys: We have just rented a server and setup a BBS for asterisk discussions at http://bbs.us.xgforce.com feel free to join. Hope you get traffic there since these lists and chat rooms work fine. What prompted you to spend money on that? -- List Manager Network Voice Comunications, Inc. netwvcom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users sted 需要一个2000兆的免费邮箱吗？网易免费邮箱是中国最多人使用的电子邮箱。 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hwo can i manage TDM04B incoming calls
hi all i'm working in the asterisk pbx, my pbx manage good outgoing calls but when i try to dial an incoming call i got this message : Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... but i could hear no thing on my diax, i'm using TDM04B, and by my diax i can dial any number for outgoing or an other diax succesfully could any body help us regards CaraMail met en oeuvre un nouveau Concept de Sécurité Globale à partir de 1,49 euros par mois___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel make problems (long)
Aldo Bergamini wrote: The error is the same, afaik. What I can't understand is why the make is entering in the directory '/ usr/src/linux-2.6.11.4-21.7-obj/i386/default'; I am by far not expert, but I would expect it to go fiddle with a '586' directory. Just a guess, your simlink is pointing to the incorrect linux source directory. Go into /usr/src/linux and do a ls -l, see where the linux simlink is pointing to. If it's incorrect, then do a rm linux and delete it. Recreate with a ln -s /usr/src/yourlinuxversionhere Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems installing asterisk-addons
Hi Angus, I don't believe it can be the root password of mysql, I used to install the addons without even haved installed mysql server yet, I guess we need to know which platform are you working on and which version you are trying to install. Thx MAG Angus Comber wrote: Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 I have set a password for root on mysql - could that be the problem? Should I remove the password? What is easiest way to do that? Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone have experience with Asterisk under Solaris 10 X86?
I've built Asterisk from recent CVS sources on a Solaris 10 X86 box. I tweaked the makefile to get the build to run using gcc. And most recently ran into va_args problems with new code in asterisk/utils.c. It seems to run OK and register with my VoIP provider, but I'm still having trouble setting-up connections to my Sipura SPA-3000. Does anyone have some experience under S10 X86? Anything you can share? Thanks, Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems installing asterisk-addons
On Thu, 2005-07-21 at 12:19 +0100, Angus Comber wrote: Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 I have set a password for root on mysql - could that be the problem? Should I remove the password? What is easiest way to do that? You haven't got far enough for that to be a problem, that would be at runtime. Are your asterisk and asterisk-addons in sync? i.e. the same release, you're not trying to mix HEAD and stable are you? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with tdm11p
sometimes my tdm11p reads the caller id and sometimes doesnt read it and give me this : Jul 21 13:55:50 NOTICE[6284]: callerid.c:307 callerid_feed: Caller*ID failed checksum Jul 21 13:55:54 WARNING[6284]: chan_zap.c:5739 ss_thread: CallerID returned with error on channel 'Zap/4-1' what is the problem please help. Its a fairly common problem with the TDM card. Best guess is the 'missed frames' across the pci bus is impacting the callerid tones in exactly the same way that missed frames is impacting the spandsp fax application. If the callerid tones happen at the same time that frames are being missed, asterisk doesn't stand a chance of recognizing the tones used during the callerid spil. Hopefully, the missed frame issue with the TDM card will be fixed soon. Until that happens, what you see is what you get. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] attended transfert
hi i would lke implement attended transfert (or consultative transfer) on asterisk server, but i don't find doc about this. Could you help me with some doc about attended transfert? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to send Fax from Asterisk
On Jul 21, 2005, at 12:00 AM, Vic wrote:Dear all, I had Tom Rymes and several others suggest how I can implement sending fax using Asterisk. The idea is to have On-Demand-Fax. Unfortunately, I wrote down the wrong workflow: the real one is: 1. Person will call our phone number 2. He will be asked to press 1 for Office 1 map, 2 for Office 2 map and 3 for Office 3 map. 3. User presses 1. 4. User is asked to enter his phone number.5. User enters his phone number and hangs up.6. Asterisk calls the number entered by user and sends a fax.Can it be done?Yes. My earlier suggestion still stands:1.) Install Hylafax and connect it with a T1 or analog modem, depending on your volume needs. You should not send hylafax calls through your asterisk server, AFAICT.2.) Set up the extension in your dialplan to do what you mention above, take the appropriate map file (in TIF format, or PS) and the entered phone number and then call Hylafax's sendfax program from your dialplan with the correct map filename and the entered phone number as arguments.3.) Lather, rinse, repeat if needed.Tom___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A104c vs. A104u
On Friday 15 July 2005 21:12, Mike M wrote: I'm just trying to decide if the extra ?200 for the A104u is worth it :) Isn't it the other way around? c u? Yes you're quite right. I think I must have just taken the headstaggers last Friday :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Working with an ongoing call
1) send sound to the caller of an ongoing call 2) retain control so the call can be terminated based on a timer (or whatever) Any tips would be greatly appreciated! Thanks in advance. Use the manager API to terminate the call if their credit reaches zero, connect and process active channels on an regular basis (as needed), use the AGI to reduce the credit by the needed amount at the end of the call (from h extension, or g option to Dial). Thanks Adam. This helps some, but I'm still not sure how you mean for me to acheive 1). I would have to perform a Dial-command no matter what, so I guess I would have to make an interruption from the manager API, but I don't manage to find a command that will acomplish that. Regards -- Eivind Trondsen People are destined to be party animals, and the technology will follow - Linus Torvalds ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue issues: timeout and leastrecent strategy
Hi, I've configured a queue with dynamic agents and leastrecent strategy. If the "least recent agent" doesn't pick up the current call from the queue, the call will be presented to him again and again, even when there's yet another agent available. I would expect that after timeout occurs on the first agent, the next to least recent agent will be tried and so on and so forth... (as it happens in case of an busy least recent agent). Did I miss something in the config or is this the intended behaviour? Thanks! cheers Jörg Nutzen Sie Ihr Postfach als virtuelle Festplatte! - Jetzt installieren! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to Digium 729
On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote: As I understand it, adding VAD/Silence would require redesigning the entire RTP stack of Asterisk. My understanding is that with the new jitter buffer both of these things are completely doable now since nothing's timed off the incoming stream... -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Working with an ongoing call
Adam, That's an interesting approach. I have a general question that arose from your comment. You suggest using the h extension (or g option of Dial) to reduce credit. What would happen if asterisk is restarted or crashes with ongoing calls? Is there any trace of those calls in order to reduce the time used? I assume you can't always guarantee all calls will be handled gracefully. Thanks, Waldo On Jul 21, 2005, at 9:04 AM, Eivind Trondsen wrote: 1) send sound to the caller of an ongoing call 2) retain control so the call can be terminated based on a timer (or whatever) Any tips would be greatly appreciated! Thanks in advance. Use the manager API to terminate the call if their credit reaches zero, connect and process active channels on an regular basis (as needed), use the AGI to reduce the credit by the needed amount at the end of the call (from h extension, or g option to Dial). Thanks Adam. This helps some, but I'm still not sure how you mean for me to acheive 1). I would have to perform a Dial-command no matter what, so I guess I would have to make an interruption from the manager API, but I don't manage to find a command that will acomplish that. Regards -- Eivind Trondsen People are destined to be party animals, and the technology will follow - Linus Torvalds ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] kphone Asterisk CVS HEAD: no audio
Dear Asterisk experts, I've just downloaded Asterisk CVS version (since I'd like to manage its configuration from RealTime). Next, I have kphone on the same Linux host, and VmWare virtual machine with Windows and X-Lite IP phone inside. I successfully tested the demo's with X-Lite, but failed to hear something with kphone (v4.1.1). There were NO problem with this kphone and stable 1.0.7 and 1.0.9 asterisk versions. Asterisk does not claim that something wrong, it logs on its condole that it just -- Playing 'demo-congrats' (language 'en'), nothing else. On the other hand, kphone finishes their log with that: = ... res_search: NO result ! res_search: NO result ! SipClient: Sending to '127.0.0.1:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for output CallAudio: Sending to remote site 127.0.0.1:13998 ERROR: Open Failed ** audioIn: openDevice Failed. CallAudio: Creating OSS-RTP Diverter dtmfsenderTimeout DspAudio: Broken pipe (b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b)(b) ... = (The complete log are attached to the e-mail.) So instead of audio I see the repeated (b) sequence dumped to the terminal from which kphone was launched. I'd blame kphone for that, but again, why I didn't experience that with the stable asterisks? Thank you a lot for any help! -- Best regards, Timur Elzhov $ kphone [1] 29730 $ Found 1 interfaces. SipClient: Listening UDP on port: 5062 SipClient: Our address: 127.0.0.1 KCallWidget: Switching calls... CallAudio: listening for incomming RTP UDPMessageSocket: Listening on 32809 UDPMessageSocket: Retrying... UDPMessageSocket: Listening on 32810 CallAudio: Opening OSS device /dev/dsp for Input and Output ERROR: Open Failed ** audioOut: openDevice Failed. CallAudio: Creating RTP-OSS Diverter SipClient: Sending: 11:22:24.494 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3 CSeq: 7312 INVITE To: sip:[EMAIL PROTECTED] Content-Type: application/sdp From: Timur Elzhov sip:[EMAIL PROTECTED];tag=6873C9D3 Call-ID: [EMAIL PROTECTED] Subject: sip:[EMAIL PROTECTED] Content-Length: 222 User-Agent: kphone/4.1.1 Contact: Timur Elzhov sip:[EMAIL PROTECTED]:5062;transport=udp v=0 o=username 0 0 IN IP4 127.0.0.1 s=The Funky Flow c=IN IP4 127.0.0.1 t=0 0 m=audio 32810 RTP/AVP 0 97 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 res_search: NO result ! res_search: NO result ! SipClient: Sending to '127.0.0.1:5060' SipClient: Receiving message... SipClient: Received: 11:22:24.556 - SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3 From: Timur Elzhov sip:[EMAIL PROTECTED];tag=6873C9D3 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 7312 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SipCall: Incoming response SipTransaction: Incoming Response SipCallMember: localStatusUpdated: 100 SipClient: Receiving message... SipClient: Received: 11:22:25.516 - SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3 From: Timur Elzhov sip:[EMAIL PROTECTED];tag=6873C9D3 To: sip:[EMAIL PROTECTED];tag=as4ee16e14 Call-ID: [EMAIL PROTECTED] CSeq: 7312 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 201 v=0 o=root 29731 29731 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 13998 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - SipCall: Incoming response SipCall: Checking for Contact and Record-Route SipCall: Setting Contact for this Call Member SipTransaction: Incoming Response SipClient: Sending: 11:22:25.523 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK5F746FD3 CSeq: 7312 ACK To: sip:[EMAIL PROTECTED];tag=as4ee16e14 From: Timur Elzhov sip:[EMAIL PROTECTED];tag=6873C9D3 Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.1.1 Contact: Timur Elzhov sip:[EMAIL PROTECTED]:5062;transport=udp res_search: NO result ! res_search: NO result ! SipClient: Sending to '127.0.0.1:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for
Re: [Asterisk-Users] Enter numeric value to use as a parameter
Here is a snippet from my remote voicemail application where a user
needs to enter a code which is then matched against the db
;
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID}) ;just so I can see who
called, may wish to save sometime
;exten = s,4,noop()
exten = s,4,DigitTimeout(1);1 second between digits
exten = s,5,SetVar(officecode=)
exten = s,6,Background(after-the-tone)
exten = s,7,Background(please-enter-your)
exten = s,8,Background(office-code)
exten = s,9,Background(vm-then-pound)
exten = s,10,Background(beep)
exten = s,11,ResponseTimeout(10);10 seconds to start dialing
exten = s,12,WaitExten
;
;Collect digits here
exten = _x,1,SetVar(officecode=${officecode}${EXTEN})
exten = _x,2,GoTo(s,11)
;
;Done collecting, check response
exten = #,1,DBget(cust=rvm/${officecode})
exten = #,2,Playback(auth-thankyou)
exten = #,3,GoTo(${cust},1112,1) ;Correct entry, goto
customer
exten = #,102,Playback(wrong-try-again-smarty)
;exten = #102,Playback(you-dialed-wrong-number)
exten = #,103,GoTo(s,5);Incorrect entry,
start over
;
On Jul 20, 2005, at 3:39 PM, Poul Møller Hansen wrote:
Can anyone please tell me how I can enter a 6 digits value to use
as a parameter in a url called by curl ?
The problem is not the curl setup, but how do I make the setup in
extensions.conf, so I can retrieve the number the users enters ?
Thanks
Poul
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Re: [Asterisk-Users] Problems installing asterisk-addons
My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j It is a version put together by Junghanns.net - for working with their ISDN cards. Mmm I wonder if that is the problem? If so then what version of asterisk-addons do I install. I didn't see anything about asterisk-addons on the junghanns.net site. Angus - Original Message - From: Dave Cotton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 21, 2005 12:56 PM Subject: Re: [Asterisk-Users] Problems installing asterisk-addons On Thu, 2005-07-21 at 12:19 +0100, Angus Comber wrote: Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 I have set a password for root on mysql - could that be the problem? Should I remove the password? What is easiest way to do that? You haven't got far enough for that to be a problem, that would be at runtime. Are your asterisk and asterisk-addons in sync? i.e. the same release, you're not trying to mix HEAD and stable are you? -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping call
Title: Dropping call Hi, after upgrading from 1.0.7 to 1.0.9 I now seem to have a call drop problem. It mostly happens after about 1min 30 secs but also happens are random intervals. Everything was fine with 1.0.9 and I'm using the same config files. Could it be a zaptel problem? Does anyone have any ideas? Regards Lee -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.9.2/54 - Release Date: 21/07/2005 ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems installing asterisk-addons
On Thursday 21 July 2005 15:28, Angus Comber wrote: My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j It is a version put together by Junghanns.net - for working with their ISDN cards. Mmm I wonder if that is the problem? If so then what version of asterisk-addons do I install. I didn't see anything about asterisk-addons on the junghanns.net site. You are right, that is the problem. I wasn't able to compile the addons with the version from junghanns.net. I suspect that it's because those addons compile the MySQL realtime extension and the Asterisk version coming with the bristuff package has no support for the realtime extension yet. Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Working with an ongoing call
On Thu, 2005-07-21 at 09:22 -0400, Waldo Rubinstein wrote: On Jul 21, 2005, at 9:04 AM, Eivind Trondsen wrote: 1) send sound to the caller of an ongoing call 2) retain control so the call can be terminated based on a timer (or whatever) Any tips would be greatly appreciated! Thanks in advance. Use the manager API to terminate the call if their credit reaches zero, connect and process active channels on an regular basis (as needed), use the AGI to reduce the credit by the needed amount at the end of the call (from h extension, or g option to Dial). That's an interesting approach. I have a general question that arose from your comment. You suggest using the h extension (or g option of Dial) to reduce credit. What would happen if asterisk is restarted or crashes with ongoing calls? Is there any trace of those calls in order to reduce the time used? I assume you can't always guarantee all calls will be handled gracefully. Well, I almost said it, but I figured by extrapolation people might work it out by themselves... Since you are checking the calls in progress on a regular basis, you might as well deduct the credit from the account on a regular basis as well. Then at completion of the call, the h extension simply charges for the time that hasn't yet been charged. Or, your software that watches and listens to the manager API slowly reduces the credit (say every 30 seconds) and then when it sees the call end (or asterisk crash) it charges the last portion of the call... Thanks Adam. This helps some, but I'm still not sure how you mean for me to acheive 1). I would have to perform a Dial-command no matter what, so I guess I would have to make an interruption from the manager API, but I don't manage to find a command that will acomplish that. Either use the dial parameters to play a sound on a regular basis, eg, you could play $UNIQUEID every 30 seconds, then your other software which is watching the manage interface and deducting money every 30 seconds can change the content of that file once you want the user to start to hear something different see the L option to dial. Alternatively, you need to be more creative and put the two calls into a meetme conference, then add your third channel (see the local channel driver) which simply plays whatever audio you need (or in fact this could be some AGI/etc)... These are just some comments I felt like making, I've never had to do this, and this is not necesarily how I would do it if I did need to (ie, if I was being paid to do this, I'd think about it more before implementing it, but for now, I can shoot my mouth off without any concern of needing to deliver on what I've said can be done). Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems installing asterisk-addons
On Thu, Jul 21, 2005 at 02:28:50PM +0100, Angus Comber wrote: My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j It is a version put together by Junghanns.net - for working with their ISDN cards. Mmm I wonder if that is the problem? If so then what version of asterisk-addons do I install. I didn't see anything about asterisk-addons on the junghanns.net site. Here asterisk-addons 1.0.9 (actually: exactly the same as 1.0.8) builds just fine with asterisk-1.0.9-bristuff-0.2.0-rc8[hj] -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues Messages not Playing
Dear friends,
Ihave a asterisk-1.0.9 verison with me in redhat linux 9.0
I am trying for ACD
I have two agents 1001,1002 and one queue called "queue1"
My requirement is like when ever any member try to enter into the queue
it should say messages like you are next and also hold time
the related sound files were present in the var/lib/asterisk/sounds
my conf files are like this
cat sip.conf[general]dtmfmode=rfc2833context=defaultport=5060disallow=allallow=ulawcontext=defaultbindaddr=192.168.68.24;nat=yes
[1000]type=friendcontext=defaulthost=dynamicnat=yes
[1001]type=friendcontext=defaulthost=dynamicnat=yes
cat extensions.conf[default]exten = 27,1,AgentLogin(1000)exten = 28,1,AgentLogin(1001)
exten = 29,1,Queue(queue1)
cat queues.conf[queue1]member = Agent/1000;rk1member = Agent/1001;rk2announceholdtime=yesannounce-frequencty=20queue-youarenext = "queue-youarenext" ; ("You are now first in line.")queue-thereare = "queue-thereare" ; ("There are")queue-callswaiting = "queue-callswaiting" ; ("calls waiting.")queue-holdtime = "queue-holdtime" ; ("The current est. holdtime is")queue-minutes = "queue-minutes" ; ("minutes.")queue-thankyou = "queue-thankyou" ; ("Thank you for your patience.")
cat agents.conf[agents]autologoff=15musiconhold = defaultgroup=1agent = 1000,4321,r1agent = 1001,4321,r2wrapuptime=5
with regards
RK
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Re: [Asterisk-Users] Problems installing asterisk-addons
On Thu, 2005-07-21 at 14:28 +0100, Angus Comber wrote: My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j It is a version put together by Junghanns.net - for working with their ISDN cards. Mmm I wonder if that is the problem? If so then what version of asterisk-addons do I install. I didn't see anything about asterisk-addons on the junghanns.net site. I use HEAD but a good starting point in your case may be asterisk-addons-1.0.9 -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems installing asterisk-addons
On Thu, Jul 21, 2005 at 03:44:03PM +0200, Christoph Eicke wrote: On Thursday 21 July 2005 15:28, Angus Comber wrote: My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j It is a version put together by Junghanns.net - for working with their ISDN cards. Mmm I wonder if that is the problem? If so then what version of asterisk-addons do I install. I didn't see anything about asterisk-addons on the junghanns.net site. You are right, that is the problem. I wasn't able to compile the addons with the version from junghanns.net. I suspect that it's because those addons compile the MySQL realtime extension and the Asterisk version coming with the bristuff package has no support for the realtime extension yet. 1.0.9 has no support for realtime yet, both in addon in in the main distribution. You seem to be mixing 1.0 and HEAD. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming MP3's from Asterisk with Ices
On 7/20/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 18, 2005 at 05:44:17PM -0700, Geoff Karl wrote: Being that my end goal is to stream an mp3 file any ideas on how this should be configured. Why stream an mp3 file in the first place? Is the network saurated? Do you really need the quality that mp3 offers you, just so you can transcode it to phone quality and waste CPU in the process? I wonder if it would be useful to stream music from another server using simply asterisk or a similar voip server: a client holds a permanent connection somewhere and provides a stream of sound. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is I would like a client to be able to listen to a meetme conference without the need of any VOIP software. I think most people have the MP3 codec installed on their local machine, but they don't have OGG installed. Do you have other ideas on how this could be done? thanks, Geoff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to send Fax from Asterisk
On Thu, 2005-07-21 at 13:00 +0900, Vic wrote:
Dear all,
I had Tom Rymes and several others suggest how I can implement sending
fax using Asterisk. The idea is to have On-Demand-Fax.
Unfortunately, I wrote down the wrong workflow: the real one is:
1. Person will call our phone number
2. He will be asked to press 1 for Office 1 map, 2 for Office 2 map
and 3 for Office 3 map.
3. User presses 1.
4. User is asked to enter his phone number.
5. User enters his phone number and hangs up.
6. Asterisk calls the number entered by user and sends a fax.
Can it be done?
Thanks,
Vic
Sure, quite easily. Setup said menus that direct through the dialplan
and terminate at something as simple as a System() that executes a
simple shell script that will then create a .call file
in /var/spool/asterisk/outgoing/, specifying the outgoing channel to use
and Application: TxFax. This is mine basically, to give you a start.
exten = x,1,System(/usr/local/bin/asterisk-sendfax ${FAXMACHINE}
${FAXFILE} ${LOCALSTATIONID}/var/spool/asterisk/outgoing/${UNIQUEID}.call)
#/usr/local/bin/asterisk-sendfax ${FAXMACHINE} ${FAXFILE}
${LOCALSTATIONID}
echo Channel: $1
MaxRetries: 0
WaitTime: 20
Context: incoming-fax
Application: TxFax
Data: $2|caller
SetVar: LOCALSTATIONID=$3
This was just a quick and dirty hack I made to try out TxFax and it
works. Just a programming note: as stated on the wiki and in various
docs, it's unwise to output straight into a call file due t timing with
when Asterisk may read it and it isn't finished being written, thus it
is always wisest to output to a temporary directory, then perform a mv
operation to place the whole file at once in the directory (not to
mention, its presence isn't written to the filesystem until all the data
is written).
Good luck!
--
-Bryce
[EMAIL PROTECTED]
NOTICE: The views expressed in this e-mail do not neccesarily reflect
those of my employer, this company, or its employees. This is a personal
e-mail and as such, the opinions expressed are my own.
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[Asterisk-Users] Asterisk, tdm card and BT line:
I dont know if is a common problem but what Ive found: First my config: Zaptel.conf: defaultzone=uk fxoks=1-2 fxsks=3-4 loadzone = uk Zapata.conf [channels] language=en context=from-pstn usedistinctiveringdetection=no usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no callwaiting=yes usecallingpres=no callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 relaxdtmf=yes rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no amaflags=billing accountcode=BT adsi=yes busydetect=yes busycount=7 hanguponpolarityswitch=yes faxdetect=both musiconhold=default signalling=fxs_ks callerid=asreceived channel=4 signalling=fxs_ks callerid=asreceived channel=3 If I call the pstn line which is on FXO port, I get caller id and it detects hungup ONLY before I run the dialparties.agi and while is running the dialparties.agi, the hungup is not detected anymore. Why ??? So is a bug in dialparties.agi or in AGI or where is the problem ?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues and timeouts
I've got several agents on a queue. However, they often forget to go
not ready or log off when they can't answer the phone.
I would like a person calling my queue to be on the queue for a max of 2
minutes, and I'm using the rrmemory strategy.
I put a timeout of 12 on the call to my agent in the [AgentQ] context
(they log on using Agentlogincallback). It all seems to work ok, except
that I get a load of pbx.c: Timeout, but no rule 't' in context
'AgentQ' in the error log. What would I use in the 't' rule to stop
this error from ocurring ?
/* extensions.conf portion for calling agent */
...
[AgentQ]
exten = _6XXX,1,Dial(SIP/${EXTEN},12)
...
/* extensions.conf portion for calling the q */
...
[macro-callq]
exten = s,1,Answer()
exten = s,2,GotoIfTime(${ARG4},${ARG5},*,*?s,4)
exten = s,3,Goto(s,6)
exten = s,4,Playback(${ARG3})
exten = s,5,Queue(${ARG2},nt,,,120)
exten = s,6,Voicemail(su${ARG1})
exten = s,7,Hangup()
...
Julian.
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Re: [Asterisk-Users] Alternatives to Digium 729
Andrew Kohlsmith wrote: On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote: As I understand it, adding VAD/Silence would require redesigning the entire RTP stack of Asterisk. My understanding is that with the new jitter buffer both of these things are completely doable now since nothing's timed off the incoming stream... -A. I figured timing could be done off a zap card or USB, just like with meetme. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New voiceovers for Allison Smith: submit today
Unfortunately, I do not have the correct pronounciations- but there are some sounds missing in say.c, for at least Portuguese: pt-ah.gsm pt-ao.gsm pt-de.gsm pt-e.gsm pt-ora.gsm pt-meianoite.gsm pt-meiodia.gsm pt-sss.gsm From what I can tell, they've been missing from the main repository for a few years, yet have been referenced in say.c for quite a while. Since I'm not a native portuguese-speaker, I'm entirely the wrong person to give a pronounciation gude here- but perhaps one of the Brazillian subscribers can? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: 20 July 2005 23:55 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] New voiceovers for Allison Smith: submit today I'm sending in a set of voiceover requests to Allison Smith this afternoon. I haven't kept up with the -users list to know if there is someone keeping track of this stuff any more... We only have a few phrases for her to record, and if anyone has applications which require Allison's voice for the asterisk-sounds repository, let me know. I'll be sending this in around 22:00 PDT today, so act fast. Please format the requests in the style: %filename%text-to-speak example: %auth-incorrect.gsm%Login incorrect. Please enter your password followed by the pound key. Any pronunciation keys should be in-line, inside of [brackets]. Please email directly to me. JT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Quit Registering with Broadvoice
I actually had the same problem for a while. It would stop registering or it would say something about register timeout. I have made two changes that have successfully resolve this issue for more then a week now. I have added nat = yes in sip.conf under broadvoice peer section and I have started using ntp on the asterisk machine ntp.nasa.gov This solved my problems. I actually have two lines registered with them. Hope that helps. Thanks, Yoni.On 7/19/05, JD Austin [EMAIL PROTECTED] wrote: Joe McConnaughey wrote: Hello - I've been using Broadvoice with Asterisk for a couple of months with no issues. Today, it has stopped registering. The Sip Debug from CLI is below. It tries to register five times and then gives up. Any suggestions? As you might suspect, I have not been able to get Broadvoice on the phone and usually get cut off after being on hold about 5 minutes. I would be VERY surprised if it was your setup that was the issue. I recently dumped them as a provider after several months of 'iffy' service. Check their BOYD setup page for asterisk in case they've changed something there, other than that crossing your fingers will do as much good as trying to contact them. JD -- JD AustinTwin Geckos Technology Services LLCemail: [EMAIL PROTECTED] http://www.twingeckos.comphone/fax: 480.288.8195 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://cigan1.smugmug.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Last two digits getting cut off?
Rob Engstrom wrote: We've just setup our [EMAIL PROTECTED] server, with our quad port card. Everything works well so far. One thing I notice is that when I leave the handset on the hook and dial a #, all is well. If I pick up the phone and dial, it cuts off at 10 digits, which is a problem if I need to dial 1+area+phone # (12 digits). The phones are Poly Soundpoint IP 600's. I'm wondering if I've missed a config to allow more than 10 digits? SIP devices (at least the good ones like the Polycoms, Cisco, SIPura, etc) have a dialplan configured on the phone. For Polycoms you do this in sip.cfg (the polycom config file) and set the dialplan.digitmap option. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one-way IAX trunking
To answer my own question... the solution is to have both ends run the same version. Mark Mark Willis wrote: Two asterisk servers, one running a recent HEAD, the other 1.0.9. I have both ends set up with trunk=yes, notransfer=yes, type=friend. I notice that the trunking works from HEAD to 1.0.9 only (the direction in which calls are originated). I know this by bandwidth usage and by iax2 trunk debug. I did have to use trunktimestamps=no on the HEAD end to keep it quiet. I assume this is the new jitterbuffer code. I know I should just upgrade the remote, but that option is difficult currently. Does anyone know why the return leg doesn't trunk? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe Enter Exit Sounds
Has anyone attempted to change the MeetMe enter and exit sounds. I see that the raw values in the enter.h and exit.h files. If I want to change the sounds is it as easy as converting the auto files to .raw and place the text in the file? I don't believe there is a header in the raw format. Thanks Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and IP500 / IP600 Boot RoM
Hello, Does anybody have the latest Boot ROMs for the IP500 and IP 600 Polycom phones. I have one of each and can't find the Boot ROM v 3 anywhere to download. I would also love a good sample phone.cfg and sip.cfg files from an Aussie asterisk user to look at. Also the ip500 is having problems trying to load the bootrom 2.6.2 ? Any ideas? Kind regardsMichael FelderIT Medic Australia Pty. Ltd.P: 03 9557 2213F: 03 9557 2214M: 0419 568 217E: [EMAIL PROTECTED]http://www.ITMedic.com.auKeeping your computer systems healthy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to Digium 729
On Thursday 21 July 2005 10:32, Matthew Boehm wrote: I figured timing could be done off a zap card or USB, just like with meetme. There's no need for a hardware timing source. The kernel timers are more than adequate for 20ms. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Did anyone else get spammed by GIZMO?
Got an email this morning with the subject Welcome to Gizmo Project. I didn't sign up with those yokels. Anyone else got spammed by them? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] a ne pas voir
Go ahead, create an MSN group. You'll be very lonely over there. -Original Message- From: ali kia [mailto:[EMAIL PROTECTED] Sent: Thursday, July 21, 2005 5:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] a ne pas voir hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Business Edition
-Original Message- From: Lee Howard [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 20, 2005 11:57 PM Subject: Re: [Asterisk-Users] RE: Business Edition Any consultant, business, or person that intends to reliably sustain ... As for the dual-license issue... there are businesses out there that may ... understand this, and I think that it's only natural. What I don't understand, though, is why the community's gratitude towards Digium should be anything more than what Digium's benevolence was towards them. normal. BUT, what is this? My contribution will not be accepted without a royalty-free disclaimer for Digium to use my work without compensation in their proprietary-licensed fork. This is what I do not like. or nominal amount. Or at least trade me in work. Give me something back of similar value... How many lines of code have you contributed to asterisk? How many lines of code were there when Digium GPL'd it? - or - Are you using Asterisk in a production environment? If yes, how much would a commercial solution have cost you? How much $$/time do you have in contributing code to asterisk? Who is getting the better end of the deal? By OS'ing Asterisk, Digium has given many folks the means to earn a living -- there are independent consultants, integrators, installers, calling card and VOIP businesses all built around Asterisk. YOU are getting some reward out of it, be it monetary or otherwise. If you contribute, others will benefit, just as you benefit from others' contributions. And Digium will sell the ABE and will have the cashflow to support the infrastructure -- mailing lists, cvs, and their full-time asterisk programmers... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Working with an ongoing call
Thanks for expanding on this. It is very clear now.WaldoOn Jul 21, 2005, at 9:47 AM, Adam Goryachev wrote:Well, I almost said it, but I figured by extrapolation people might work it out by themselves... Since you are checking the calls in progress on a regular basis, you might as well deduct the credit from the account on a regular basis as well. Then at completion of the call, the h extension simply charges for the time that hasn't yet been charged. Or, your software that watches and listens to the manager API slowly reduces the credit (say every 30 seconds) and then when it sees the call end (or asterisk crash) it charges the last portion of the call... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call queue advice
Hi all, Looking for some advice as to how best to configure a call queue situation. Basically wondering whether it can be achieved with a standard asterisk based queue or not? I have calls coming in to several different DDIs, these would in theory all route to same call queue as its important that all calls are handled in the order they are received. However, I have now learnt that certain agents should not take certain calls. Is there anyway that I can make these agents unavailable as far as the queue is concerned for these specific DDIs? Any help would be really appreciated. One of the main reasons for wanting to keep using queues.conf is that I had wanted to use agentlogin. Cheers, Neil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to Digium 729
Andrew Kohlsmith wrote: On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote: As I understand it, adding VAD/Silence would require redesigning the entire RTP stack of Asterisk. My understanding is that with the new jitter buffer both of these things are completely doable now since nothing's timed off the incoming stream... ...when the new jitterbuffer is included and if it's enabled... Please help us test the SIP/RTP jitterbuffer! It's available in the bug tracker! /Olle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF not working
Hi all, I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no longer works with external phone systems. I have a Wildcard TDM400P with 4 FXO's? (it connects to analog lines). No changes were made to the config files. Here's my config: /etc/zaptel.conf fxsks=1-4 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes rxgain=2.0 txgain=2.0 callgroup=1 pickupgroup=1 musiconhold=default context=incoming group=1 signalling=fxs_ks echocancel=64 echocancelwhenbridged=yes relaxdtmf=yes channel = 1-3 [pete_desk] ;Pete's Desk phone (Polycom IP 300) type=friend username=pete_desk secret=pass context=longdistance callerid=Pete 601 host=dynamic mailbox=601 dtmfmode=inband disallow=all allow=ulaw allow=alaw Thanks, Pete ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] attended transfert
attended transfer are implemented on some cases on the phone side, if you need attended transfers on dial plan you need use asterisk CVS HEAD, i are using asterisk CVS HEAD and attended transfer work very well. just install asterisk CVS HEAD and configure features.conf file, on voip-info.org have good example of features.conf On 7/21/05, sylvain garcia [EMAIL PROTECTED] wrote: hii would lke implement attended transfert (or consultative transfer) onasterisk server,but i don't find doc about this.Could you help me with some doc about attended transfert?thanks___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- David Romero ROMDAV## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] a ne pas voir
you can download amsn it work under linux i have it and it works succesfully De: Christoph Eicke [EMAIL PROTECTED] A: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Objet: Re: [Asterisk-Users] a ne pas voir Date: Thu, 21 Jul 2005 13:13:59 +0200 On Thursday 21 July 2005 12:41, ali kia wrote: hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group also I would prefer not to switch to something M$ based... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com a href=http://lists.digium.com/mailman/listinfo/asterisk-usershttp://lists.digium.com/mailman/listinfo/asterisk-users/aTo UNSUBSCRIBE or update options visit: a href=http://lists.digium.com/mailman/listinfo/asterisk-usershttp://lists.digium.com/mailman/listinfo/asterisk-users/a Protek-on: CaraMail met en oeuvre un nouveau Concept de Sécurité Globale - www.caramail.com___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HOW TO RECEIVE FAX
Hi all!i'm in course to implement Faxing in my
asterisk box and for that I've installed all succesfully like libtiff and
spandsp and next rebuild and reinstall asterisk modules, but when i call to
Rxfax from dialplan nothing happens and i get some errors like "XCN with final
frame tag In state 9" or a papercopy of a transmision report from the fax
machine with COMMUNICATION ERROR CODE 41 error. My Dialplan shows like this over
a Zap channel configuration:
[macro-faxreceive] (called from a main dialplan
script)exten = s, 1, Answerexten = s, 2,
SetVar(FAXFILE=/var/spool/asterisk/fax/in/${UNIQUEID}.tif)
exten = s, 3, rxfax(${FAXFILE})
So, what i need to send succesfully Faxes from any
Fax machine to Asterisk?...I have reading all about fax and asterisk and
dont help mewith my implementation.
What are the right configuration parameters that
would show my ZAPATA.CONF file to receive faxes?
Thanks for any guide to solve this.
G.
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Re: [Asterisk-Users] Asterisk and flash disks
Paul Hewlett wrote: On Wednesday 20 July 2005 15:49, Angus Comber wrote: Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and deleted immediately? I have been using flash-based asterisk and it works fine as long as you are careful about the maker - SanDisk did not work properly, we currently use the RiData flash disk and are happy with it. I have a Kingston Tech flash disk that was flaky and finally failed yesterday. They are not the fastest - running hdparm -tT on them reveals a speed of 2Mb/s which is about a third of the speed of 100Mbits ethernet. For call recording I usually add an IDE hard drive and make sure that most filesystems (e.g. /var,/tmp..) are loaded into a RAM disk Paul Hewlett Paul, SanDisk CF cards are often considered to be the best around. What problems were you having? CF is both slow and fast. Seek times are very low, but sustained data transfer rates are not very good (ESPECIALLY for writes). 2Mb/s is actually more like 1/4 - 1/5 the speed of 100mbps ethernet... Are you aware that I have created a distro specifically for running Asterisk from compact flash? Perhaps you should take a look: http://www.astlinux.org -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] a ne pas voir
On Thu, 2005-07-21 at 15:56 +, ali kia wrote: you can download amsn it work under linux i have it and it works succesfully I don't think the software was the point, it was the hotmail part. Never mind it would be a bit like Esperanto, if you can find the other person who speaks it you can have a conversation. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to Digium 729
Olle E. Johansson wrote: Andrew Kohlsmith wrote: On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote: As I understand it, adding VAD/Silence would require redesigning the entire RTP stack of Asterisk. My understanding is that with the new jitter buffer both of these things are completely doable now since nothing's timed off the incoming stream... ...when the new jitterbuffer is included and if it's enabled... Please help us test the SIP/RTP jitterbuffer! It's available in the bug tracker! /Olle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Is the May 4 version the newest? http://www.astertest.com/downloads/sip jitter buffer/latest/ Best regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to Digium 729
Olle E. Johansson wrote: ...when the new jitterbuffer is included and if it's enabled... Please help us test the SIP/RTP jitterbuffer! It's available in the bug tracker! /Olle post 'da bug number -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error while writing audio data: : Broken pipe ... segmentation fault
Hi, My asterisk server crashed with that error message. I'm using 1.0.9. I don't know how to replicate the problem (although this is the second time the server crashes). I have two (~40M each) core files, but I don't know how to debug. any help is well appreciated. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW TO RECEIVE FAX
How are these faxes arriving in your * server?
If you have a line handler card like the TE400 types then this should
not be a problem. If you are getting your faxes from a VoIP service like
Broadvoice then you should make sure that you are usling ULAW as the
codec. Anything else will conpress the fax tones and when uncompressed
by your * server they will not work.
Having said that, Ulaw may also not work. Many folks have no problems
with using ULaw for their VoIP prover codec but some folks do.
Mark
Gustavo A. Gonzalez wrote:
Hi all!i'm in course to implement Faxing in my asterisk box and for that I've
installed all succesfully like libtiff and spandsp and next rebuild and reinstall
asterisk modules, but when i call to Rxfax from dialplan nothing happens and i get some
errors like XCN with final frame tag In state 9 or a paper copy of a
transmision report from the fax machine with COMMUNICATION ERROR CODE 41 error. My
Dialplan shows like this over a Zap channel configuration:
[macro-faxreceive] (called from a main dialplan script)
exten = s, 1, Answer
exten = s, 2, SetVar(FAXFILE=/var/spool/asterisk/fax/in/${UNIQUEID}.tif)
exten = s, 3, rxfax(${FAXFILE})
So, what i need to send succesfully Faxes from any Fax machine to
Asterisk?...I have reading all about fax and asterisk and dont help me with my
implementation.
What are the right configuration parameters that would show my ZAPATA.CONF file to receive faxes?
Thanks for any guide to solve this.
G.
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--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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Re: Re: [Asterisk-Users] a ne pas voir
On Thu, 2005-07-21 at 15:56 +, ali kia wrote: you can download amsn it work under linux i have it and it works succesfully I think he was refering to the service provider, MSN as in MicroSoft Network, as opposed to the operating system. There is already a large enough user base and comments about the irc channel irc.freenode.net #asterisk, which is accessable via many clients in many operating systems and even via web browsers if you have a server with the appropriate software in place, or applets local to your system. To make the MSN chat meaningful you would have to get a bunch of people to convert, and it is always much harder to get everyone to change the way they currently do things to something new unless you can prove that the new way is somehow superior. MSN does not appear to be superior to any other realtime chat network, so it will be a tough sell. You can obviously go there yourself, and attempt to get others to follow you, I just see that path as a difficult one, the easiest one would be to follow the crowd and do what they are already doing, but innovators never got anywhere by following the crowd. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thailand DIDs
Anyone know of a place to get a Thailand DID that will ring in to asterisk in the US at a nice price? Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and flash disks
Hi, On Wed, 2005-07-20 at 14:49 +0100, Angus Comber wrote: I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Beware that some flash producers sacrifice seek speed for transfer speed. This is okay for most usages, but not for harddrive-usage. 16 CompactFlash Memory Cards Roundup: Professional Photographer’s Best Choice: http://www.xbitlabs.com/articles/memory/display/16-cflash-roundup.html Or just skip right to the graphs: http://www.xbitlabs.com/articles/memory/display/16-cflash-roundup_8.html /Anders ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New features for e164.org
For a long time now we've allowed people to publish a wide variety of URI against their enum records such as SIP/IAX2/H323 for VoIP and other types for non-VoIP such as HTTP/MAILTO etc. For the most part these record types aren't listed or aren't utilised so I've done up a quick hack for firefox users as a proof of concept and I'm hoping others will take advantage of this and use enum.164 records more widely. If you go to our main website (http://www.e164.org) you will see a paragraph and a link under Latest News which if you click on it will load a new search engine in the top right drop down, from there you can change to the enum.164 to website and enter your phone number and if you have any http records it will redirect your connection to the first priority. This could be expanded to a whois type listing where you see all websites/emails/other contact types and you can pick and choose which site you get redirected to. Due to request we've also added an additional record type to the system so people can list their GPG finger prints, and we're trying to work out the best way to list a postal/physical address this way people really could end up listing their name and a single number on a business card and all the other information can be referenced from our DNS zone. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thailand DIDs
Anyone know of a place to get a Thailand DID that will ring in to asterisk in the US at a nice price? Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for Thai DIDs
Anybody know where to find Thailand DIDs that can ring in to my * in the USA on SIP? Oh, and a good price, too! ;) Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to Digium 729
Matthew Boehm wrote: Olle E. Johansson wrote: ...when the new jitterbuffer is included and if it's enabled... Please help us test the SIP/RTP jitterbuffer! It's available in the bug tracker! /Olle post 'da bug number http://bugs.digium.com/view.php?id=3854 Download the patch, apply it and try it out! /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Business Edition
Jay Milk wrote: Who is getting the better end of the deal? Well, Digium, of course. I certainly hope that they've made way more money from Asterisk than I ever expect to save or make. And I certainly expect that Digium has made way more money from Asterisk because they've open-sourced it than they ever could have hoped to make by not doing so. By OS'ing Asterisk, Digium has given many folks the means to earn a living Yes, indeed, as well as themselves. And all that I'm saying is that this is completely expected, natural, and fair. I've no argument against this. What I am saying, though, is that Digium didn't give out royalty-free proprietary licenses to Asterisk, instead, they gave out GPL licenses to Asterisk. Why, then, do they require that contributions are made any differently? Why do they require freedoms with contribution that they did not give with theirs? Well, probably because they believe that they're owed that, and probably because many others in the community not unlike yourself agree with that opinion as well. Lee. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sccp new release 20050721
Please visit http://chan-sccp.berlios.de/ - New sccp.conf parser. You need to edit your old sccp.conf and update it according to the new sccp.conf (conf/sccp.conf) - Button template on phone 79[2467]0 has been improoved. Now you can choose the line/speedial button position. 7914 can now use all the buttons. For example on 7960 you can have 1 line button, 1 speedial, 1 empty button (separator), 1 line and 2 speedials :-) - Fixed a 7960 line label issue - Minor cleanups. The call park stuff is not compiled by default (and you need to load = res_features.so) . To try it: make clean CFLAGS=-DCS_SCCP_PARK make install Sergio Chersovani ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iaxy call waiting problems
All I currently have asterisk setup at home and everything seems to be working great running Asterisk v1.8 and several Iaxy devices. Call waiting signals come through to alert the user of a call waiting call but when using the flash button on the analog phone the current user is placed on hold and a new dial tone is given, the new call is never connected. I have been unsuccessful in finding any information on setting up call waiting in Asterisk. Any help or ideas that could be causing the call waiting feature to not function properly would be great. Thank you for the time in advance. Chris B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a ne pas voir
On Thu, Jul 21, 2005 at 10:41:57AM +, ali kia wrote: hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group Create one, and announce it in the proper channels, if you like. Frankly, I see no reason why you shouldn't use the channel #asterisk on Freenode. Consider using an IM software that handles both IRC and msn-messanger (like gaim). But before you do that, plese get a decent email software. The one you use (webmail from lycos(?) ) does not create message IDs, and thus decent threading for your messages is practically impossible. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Free Music
Hello, On Wed, 20 Jul 2005, Wiley Siler wrote: For the fella who wanted MOH music Royalty free stuff can be found here.. The Acoustic Guitar is a nice collection... http://www.freeplaymusic.com/ Cheers, W I spoke with Scott at freeplay today, who said that licensing is required ($25/song) unless your PBX is for home use. - Ivan. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Routing by DID
Hi, This is my setup; 1. PSTN == Cisco == Internet == Asterisks == Grandstream Phone 2. Grandstream ATA =SIP Proxy== Internet == Asterisks == Grandstream Phone In both cases above when I dialed the DID (say) 1-213-444-1234 from either the PSTN or Grandstream ATA the response I "see" on the asterisks is somthing like this below; "SIP/kkk.kkk.kkk.kkk-084cfc38" == where kkk.kkk.kkk.kkk is the IP address on the Cisco or SIP Proxy I was actually expecting something like SIP/12134441234 that will allow me the opportunity to route the incoming calls by DID to different context base on each DID. How do I achieve this? -Olusoji (Soji) OyenugaSenior VoIP Project ManagerModern Digital Communications IncPhone: 1-306-683-2089Email: [EMAIL PROTECTED]MSN: [EMAIL PROTECTED]http://www.mdci.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming MP3's from Asterisk with Ices
On Thu, Jul 21, 2005 at 07:04:43AM -0700, Geoff Karl wrote: On 7/20/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 18, 2005 at 05:44:17PM -0700, Geoff Karl wrote: Being that my end goal is to stream an mp3 file any ideas on how this should be configured. Why stream an mp3 file in the first place? Is the network saurated? Do you really need the quality that mp3 offers you, just so you can transcode it to phone quality and waste CPU in the process? I wonder if it would be useful to stream music from another server using simply asterisk or a similar voip server: a client holds a permanent connection somewhere and provides a stream of sound. I would like a client to be able to listen to a meetme conference without the need of any VOIP software. I think most people have the MP3 codec installed on their local machine, but they don't have OGG installed. Do you have other ideas on how this could be done? Provide a simple, dumbed-down iax client that will connect to your server to a specific extension. iaxclient comes with a simple command-line program that has all the functionality you need, and you just need to put some GUI around it. Alternatively, iaxcomm should be hackable. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
