[Asterisk-Users] New idea
Dear All, Thank for your visit our site, I found some users can not read our home page from some browser, I will move all the pgae to the top directory later. I had some idea, do you agree? I want to setup a voip provider group to share the local PSTN connection, every member must provide at least one T1 Pri to the group, other member can share the ports and bandwith, I think this may reduce many money at the member side, if more than one member on the same country, we will change the queue weekly, if any member can provide some server at different country, we can low the exchange traffic. We can use Asterisk PBX to do this exchange, any idea or comment? Cary LEUNG Network Operator Hong Kong VOIP Exchange Network ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 addon mysql
Tommy, Thanks, how do i get the older version of asterisk-addons? -- Harold Workman Quoting T. Chan [EMAIL PROTECTED]: Hi, I got the same thing, so what I did was for the asterisk-addons, I used CVS April instead of the most current CVS and it worked. Of course, I would have liked to use the most current CVS of asterisk-addons as well, but since the old version works with the most current version of asterisk anyways, I left it like that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: Sunday, June 27, 2004 3:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk addon mysql hi, ive read through the last few posts with people having problems compiling the asterisk-addons for mysql support, and none of them have helped me resolve my compile problem. I currently have -- CVS-06/24/04-22:20:31 and downloaded asterisk-addons. I compiled * first then asterisk-addons, have added CFLAGS+=-I../asterisk/include When I try to make install for asterisk-addons i get [EMAIL PROTECTED] asterisk-addons]# make clean ; make install rm -f *.so *.o .depend cc -fPIC -I../asterisk -D_GNU_SOURCE -I../asterisk/include -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function decla ration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 I have MySQL-server and devel upgraded at version 4.0.20 on a Fedora Core 1. I would really love to have mysql support Harold Workman This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] IAX Phone Issues/McAfee Virus Scan vs. IAX Phone
Have you ever looked into adding support for dialing directly from a browser? i.e. a href=iax:[EMAIL PROTECTED]click here to call foo/a and IAX Phone pops up and dials. I think estara's SIP softphone supports this. -Brian On Sun, 27 Jun 2004 20:49:55 -0500, Steven M. Sokol [EMAIL PROTECTED] wrote: --Request For Bug Reports-- I'm working on the next release of IAX Phone. Please let me know what, if any, issues you who use it may have run into. I hope to be able to release a new version in the next two weeks. Some fixes/features: - Conferencing - Proper handling of 'qualify' - Intercom - Paging - Phone Book --Virus Scanner Problems?-- I have been working through a number of the bugs already submitted to me. One is a rather large delay (300 ms+) for incoming audio. Some of this can be linked to the way it interfaces with the windows audio system. More of it, however, appears to be linked to Virus Scanning software. I tested on a system running McAfee 8.0. and found that the scanner software introduces the majority of the delay and causes other problems as well. The thread that runs the audio interface runs at high priority, but from time-to-time the processor spikes up due to some action on the part of the virus scanner and the audio drops out. Has anybody else experienced such issues with IAX Phone and McAfee? How about other virus scanners? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 idea
what site? - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 27, 2004 11:01 PM Subject: [Asterisk-Users] New idea Dear All, Thank for your visit our site, I found some users can not read our home page from some browser, I will move all the pgae to the top directory later. I had some idea, do you agree? I want to setup a voip provider group to share the local PSTN connection, every member must provide at least one T1 Pri to the group, other member can share the ports and bandwith, I think this may reduce many money at the member side, if more than one member on the same country, we will change the queue weekly, if any member can provide some server at different country, we can low the exchange traffic. We can use Asterisk PBX to do this exchange, any idea or comment? Cary LEUNG Network Operator Hong Kong VOIP Exchange Network ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 on 64bit ?
Nicholas Bachmann [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Dr. Rich Murphey [EMAIL PROTECTED] wrote: How do you balance the number of active connections per server? In theory, you could use a load balancer. That's as long as you can share the SIP/IAX registrations between the nodes. I'm not sure if that can be done yet - I haven't looked into it. It can. SIP registration info can be stored in a database; see http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers Sorry - I meant the information relating to registrations that have already been made. Like you get when you type sip show users. Perhaps that's not necessary anyway; The user should attempt to re-register if the connection is broken, and may find itself connecting to a new server automatically. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Cron
On Mon, 2004-06-28 at 02:02, Samantha (Femtech) wrote: Is there a cron that I con do to replace this, as the fx0 card doesnt hang up properly I had the same problem here, and fixed within zapata.conf by adding these lines: busydetect=1 busycount=5 Try reading this also: http://www.voip-info.org/wiki-Asterisk+Disconnect+Supervision and http://voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] H.323 Audio problem UPDATE
Scott Stingel a écrit : Hi- In answer to your questions: Someone on Friday had said that disabling Fast Start corrected the audio problem with H.323, so yesterday I tried to disable it in ~/asterisk/channels/h323/ast_h323.cpp. Today, I noticed that Jeremy (NuFone) uploaded a new version of this file with the same fix: Change the line: BOOL noFastStart; To: BOOL noFastStart = TRUE; Unfortunately, this made no difference for connections from my customer's Cisco 5300, so I decided to abandon the built-in h323 in favour of oh323. Maybe you'll have better luck with the original code. I updated the to the cvs-27/06/04, applied the changes above and it works. I'm not using any cisco devices but the GNUgk [...] -- dash ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 to isdn-capi call problem
anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI -- Executing Dial(SIP/102-767c, CAPI/2:5) in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making progress passing it to SIP/102-767c Jun 28 10:51:21 NOTICE[278545]: channel.c:1654 ast_set_read_format: Unable to find a path from G723 to ALAW Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: Unable to find a path from ULAW to G723 -- CAPI[contr1/2003002]/0 is ringing Jun 28 10:51:21 WARNING[278545]: chan_sip.c:1788 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 8/4) Jun 28 10:51:21 WARNING[278545]: channel.c:1485 ast_prod: Prodding channel 'SIP/102-767c' failed Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: Unable to find a path from SLINR to G723 Jun 28 10:51:21 WARNING[278545]: indications.c:76 playtones_alloc: Unable to set 'SIP/102-767c' to signed linear format (write) -- CAPI Hangingup == Spawn extension (from-sip, 9, 1) exited non-zero on 'SIP/102-767c' ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 addon mysql
cvs checkout -D mm/dd/yy asterisk-addons -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: Monday, June 28, 2004 1:03 AM To: [EMAIL PROTECTED]; T. Chan Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk addon mysql Tommy, Thanks, how do i get the older version of asterisk-addons? -- Harold Workman Quoting T. Chan [EMAIL PROTECTED]: Hi, I got the same thing, so what I did was for the asterisk-addons, I used CVS April instead of the most current CVS and it worked. Of course, I would have liked to use the most current CVS of asterisk-addons as well, but since the old version works with the most current version of asterisk anyways, I left it like that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: Sunday, June 27, 2004 3:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk addon mysql hi, ive read through the last few posts with people having problems compiling the asterisk-addons for mysql support, and none of them have helped me resolve my compile problem. I currently have -- CVS-06/24/04-22:20:31 and downloaded asterisk-addons. I compiled * first then asterisk-addons, have added CFLAGS+=-I../asterisk/include When I try to make install for asterisk-addons i get [EMAIL PROTECTED] asterisk-addons]# make clean ; make install rm -f *.so *.o .depend cc -fPIC -I../asterisk -D_GNU_SOURCE -I../asterisk/include -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function decla ration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 I have MySQL-server and devel upgraded at version 4.0.20 on a Fedora Core 1. I would really love to have mysql support Harold Workman This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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:Latest Echo changes
As you are in the UK I assume you are using the X101P like me. The best you can do with this card is compile agressive echo cancelling on and not have the tx gain too high. I hope that when the new FXO module is available here the issue will go away. Out of curiostity anychance you can list what you did? Settings etc, you say not have it too high etc, what have you got yours set to? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 audio
Upgrade your firmware on the SPA-2000 and see if it fixes the one way audio problem. I had this problem and worked with Sipura to get it resolved. If you are running a firmware earlier then version 2.0.6(c) then you will have this problem. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Mattinen Sent: Sunday, June 20, 2004 2:00 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] One way audio Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the offending line looks like this: [202] type=friend username=202 secret=voip-analog0 host=dynamic context=from-sip reinvite=no canreinvite=no disallow=all allow=ulaw nat=0 It works fine when calling between internally, or when the SPA-2000 is the calling source, but if a call comes in on a zap channel, the one way audio problem appears. -- Seth et lux in tenebris lucet Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disappointed
Well, I have to confess that I am disappointed that in a fairly high volume list like this, I haven't had one reply to the questions I've asked. (I know I haven't got any right to expect a reply, but communities are usually fairly helpful). It might be really obvious to you guys, but if you have not a lot of experience with ISDN/PBXs, it's hard to understand. I'm going to unsubscribe, so if anyone feels that they can help me out, please reply to my email address. ( calum dot asterisk **at** umtstrial dot c o dot u k ) Does this card work/can it be made to work with Asterisk? lspci: 07:06.0 Network controller: Eicon Technology Corporation DIVA Server PRI-30M 2.0 Can I establish 2 outbound calls with it, and conference them together? Thanks once again. Don't bother with flames. Calum -- Random russian saying: If the thunder is not loud, the peasant forgets to cross himself. jabber: [EMAIL PROTECTED] pgp: http://gk.umtstrial.co.uk/~calum/keys.php Linux 2.6.5-gentoo 10:06:14 up 19 days, 22:34, 1 user, load average: 0.35, 0.31, 0.29 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Cron
Samantha (Femtech) [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Is there a cron that I con do to replace this, as the fx0 card doesnt hang up properly phonegc:/home/samantha# asterisk -r Asterisk CVS-05/30/03-17:17:07, Copyright (C) 1999-2001 Linux Support You're using an Asterisk server built in 2003, and possibly an old Zaptel driver as well. Perhaps you might have more luck with your hangup problem after an upgrade. As for the cron job, you should be able to use the following: /path/to/your/asterisk -r -x restart when convenient -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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:Latest Echo changes
Cheers Chris! Any idea when the new FXO Module will be available? My setup = Grandstream/ATA186 Asterisk FXO Chris Bond [EMAIL PROTECTED] wrote: As you are in the UK I assume you are using the X101P like me. The bestyou can do with this card is compile agressive echo cancelling on and nothave the tx gain too high. I hope that when the new FXO module isavailable here the issue will go away.Out of curiostity anychance you can list what you did? Settings etc, yousay not have it too high etc, what have you got yours set to?___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself
Re: [Asterisk-Users] Re:Latest Echo changes
I have AGGRESSIVE_SUPPRESSOR uncommented in zconfig.h and txgain set to 4.0; Its a little quiet but usable. I've stopped playing with the settings now cos I hope to get the new fxo module very soon. Chris - Original Message - From: Chris Bond [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 28, 2004 10:00 AM Subject: RE: [Asterisk-Users] Re:Latest Echo changes As you are in the UK I assume you are using the X101P like me. The best you can do with this card is compile agressive echo cancelling on and not have the tx gain too high. I hope that when the new FXO module is available here the issue will go away. Out of curiostity anychance you can list what you did? Settings etc, you say not have it too high etc, what have you got yours set to? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Disappointed
Yes it is possible, with the chan_CAPI drivers from junghanns.net i only used the 4BRI cards from Eicon but they are similar to the PRI cards i didn't have any ISDN knowledge before. but first tried to install the card with CAPI on a redhat 9 machine with exactly the description from eicon then tried to start the chan_capi and it finally worked !! It took me some research as above all i was not really a linux guru and about the conferencing question = yes Michael -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Calum Verzonden: ma 28/06/2004 11:11 Aan: [EMAIL PROTECTED] CC: Onderwerp: [Asterisk-Users] Disappointed Well, I have to confess that I am disappointed that in a fairly high volume list like this, I haven't had one reply to the questions I've asked. (I know I haven't got any right to expect a reply, but communities are usually fairly helpful). It might be really obvious to you guys, but if you have not a lot of experience with ISDN/PBXs, it's hard to understand. I'm going to unsubscribe, so if anyone feels that they can help me out, please reply to my email address. ( calum dot asterisk **at** umtstrial dot c o dot u k ) Does this card work/can it be made to work with Asterisk? lspci: 07:06.0 Network controller: Eicon Technology Corporation DIVA Server PRI-30M 2.0 Can I establish 2 outbound calls with it, and conference them together? Thanks once again. Don't bother with flames. Calum -- Random russian saying: If the thunder is not loud, the peasant forgets to cross himself. jabber: [EMAIL PROTECTED] pgp: http://gk.umtstrial.co.uk/~calum/keys.php Linux 2.6.5-gentoo 10:06:14 up 19 days, 22:34, 1 user, load average: 0.35, 0.31, 0.29 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. winmail.dat
RE: [Asterisk-Users] Re:Latest Echo changes
I believe its out if you call digium direct - im gonna give them a call later see what the latest is. From: taf taffey [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 10:31 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Re:Latest Echo changes Cheers Chris! Any idea when the new FXO Module will be available? My setup = Grandstream/ATA186 Asterisk FXO
Re: [Asterisk-Users] Re:Latest Echo changes
Yes but telappliant (the uk disti) have yet to get approval for it in the UK. I've just fired of an e-mail to them as they said they should have it by the end of the month. As you say though you can go direct ... Chris - Original Message - From: Chris Bond [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 28, 2004 10:43 AM Subject: RE: [Asterisk-Users] Re:Latest Echo changes I believe its out if you call digium direct - im gonna give them a call later see what the latest is. _ From: taf taffey [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 10:31 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re:Latest Echo changes Cheers Chris! Any idea when the new FXO Module will be available? My setup = Grandstream/ATA186 Asterisk FXO ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to forward voice
Hi again, always latest CVS from 27/06/04. Calling to a SIP gateway I receive: Unable to find a path from G723 to ALAW Unable to find a path from ULAW to G723 Asked to transmit frame type 4, while native format is 1 (read/write = 8/4) Unable to forward voice [last messages repeated lot of times] Acked pending invite 102 - My phone number ... No path to translate from SIP/... to SIP/... Had to drop call because I couldn't make SIP/... compatible with SIP/... Even if I force my sip.conf to use only g723.1 I have the same result. BTW, if I want to modify my codecs in a sip context, it's not taking in account by asterisk. Is'it normal behaviour? -- dash ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Why? oh why can't I dial out?
On Sun, 27 Jun 2004 17:25:56 +0100, Vassilis Konstantinou
[EMAIL PROTECTED] wrote:
Thanks for the reply Greg,
The definition for the console is
[globals]
;CONSOLE=Console/dsp; Console interface for demo
CONSOLE=Zap/1
so if I am mistaken I have commented out the dsp and I am using Zap/1 the
X100P card. Is this ok?
the clock is 123 so dialing 9123 should get me there.
Best regards
Vassilis
At 17:12 27/06/2004, you wrote:
So, assuming that calls from your SIP device are in the same context as
the above extensions, all extensions beginning with a 9 should be dialled
on ${CONSOLE}. On my box, ${CONSOLE}=console/dsp... the sound card. Is
yours set to something similar (or is it really set to dial the zap
interface?)
Not being from the UK myself, I don't know whether the clock's number is
123 or 9123. If it's 9123, then you should be dialing 99123 in order to
get through your dialplan with the 9123 still intact to send to the PSTN.
Greg
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[Asterisk-Users] SetGroup and CheckGroup
Does anyone know if SetGroup and CheckGroup apply to only current context or is it per server based? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Compiling and Loading asterisk-oh323 0.6.2
Use the 0.6.2a version. Michael. Brian Wilkins wrote: Hi, I having a problem compiling the wrapper for oh323. I am running Debian, kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the openh323 version I have is 1.13.5. I execute the following commands first before attempting to compile the wrapper: pwlib_1.6.6: make both openh323 1.13.5 ./configure make opt asterisk-oh323 0.6.2 make I also applied the patch that is said that is needed for openh323 1.13.5. And I get the following errors: make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE - I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:660: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_oh323.c:660: initializer element is not constant chan_oh323.c:660: (near initialization for `oh323_ep_list.lock') I have been sucessful before in compiling all packages before. I still have the libraries installed from the wrapper package. I decided to try and download a newer version of openh323 and pwlib, but they did not compile correctly either, so I went back to the versions that I listed above, because I knew they would compile correctly. I still have the successfully compiled and installed modules, and before attempting to upgrade to the newer versions of pwlib and openh323, I ran asterisk -. This is the error I got : [chan_oh323.so]Jun 25 13:45:13 WARNING[1024]: loader.c:242 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: __tf6PMutex Jun 25 13:45:13 WARNING[1024]: loader.c:423 load_modules: Loading module chan_oh323.so failed! So, I am wondering what is wrong and whether the packages I have built are compatible. Any help on this is greatly appreciated. -- Brian Wilkins [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Protocol Error (6) using Zaphfc
Hei,
please never try to dial out on a particular b channel, you have to dial
out on a zaptel group which includes both b channels of the BRI line.
In a p2mp setup YOU cannot know which b channel will be chosen!
exten = _X.,1,Dial(ZAP/g1/${EXTEN})
will do(note the 'g')
best regards
Klaus
Am Mo, 2004-06-28 um 12.45 schrieb nrb:
Hi!
Has anybody seen anything like this using zaphfc?
On outgoing calls (via isdn) , the line gets hung-up as soon as the
called
party answers.
As seen below i get some protocol error (6) - but i'm not sure if this
is
related to the hang-up which apparently comes a little earlier?!
Incomming calls on the isdn (zaphfc) interface is working just fine
(P.S. what about the D-channel going up down all the time - is that
normal? )
Kind Regards
NRB
Setup
Bri-stuff - 0.0.20
Asterisk CVS-HEAD-06/23/04-15:45:48 built by
[EMAIL PROTECTED] on a
i686 running Linux
Zapata.conf:
[channels]
switchtype = euroisdn
; p2mp TE mode
signalling = bri_cpe_ptmp
; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; p2p NT mode
;signalling = bri_net
pridialplan=local
prilocaldialplan=local
echocancel=yes
immediate=yes
group = 1
context=demo
channel = 1-2
Zaptel.conf:
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
Example where a sip client (2203) is calling 7024
From Asterisk:
== D-Channel on span 1 down
== D-Channel on span 1 up
-- Executing Dial(SIP/2203-5779, Zap/1/7024) in new stack
-- Making new call for cr 135
Protocol Discriminator: Q.931 (8) len=32
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: SETUP (5)
Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer
capability:
Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Ext: 1 User information layer 1: A-Law (35)
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive
Dchan: 0
ChanSel: B1 channel
]
Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number passed network
screening
(1) '2203' ]
Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '7024' ]
Sending Complete (len= 0)
-- Called 1/7024
Protocol Discriminator: Q.931 (8) len=7
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: CALL PROCEEDING (2)
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive
Dchan: 0
ChanSel: B1 channel
]
-- Processing IE 24 (Channel Identification)
Protocol Discriminator: Q.931 (8) len=12
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: ALERTING (1)
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Network beyond the interworking point (10)
Ext: 1 Progress Description: Inband information or appropriate
pattern now
available. (8) ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Network beyond the interworking point (10)
Ext: 1 Progress Description: Unknown (1) ]
-- Processing IE 30 (Progress Indicator)
-- Processing IE 30 (Progress Indicator)
-- Zap/1-1 is ringing
Protocol Discriminator: Q.931 (8) len=15
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: CONNECT (7)
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Public network serving the remote user (4)
Ext: 1 Progress Description: Unknown (4) ]
Time Date (len= 5) [ 04-06-28 11:58 ]
-- Processing IE 30 (Progress Indicator)
-- Processing IE 41 (Date/Time)
Protocol Discriminator: Q.931 (8) len=4
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: CONNECT ACKNOWLEDGE (15)
-- Zap/1-1 answered SIP/2203-5779
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
Connect
Request
Protocol Discriminator: Q.931 (8) len=8
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: DISCONNECT (69)
Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location:
Private network serving the local user (1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]
-- Hungup 'Zap/1-1'
== Spawn extension (intern, 7024, 1) exited non-zero on
'SIP/2203-5779'
Protocol Discriminator: Q.931 (8) len=4
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: RELEASE (77)
-- Channel 1, span 1 got hangup
Protocol Discriminator: Q.931 (8) len=11
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: RELEASE (77)
Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location:
Public network serving the local user (2)
Ext: 1 Cause: Recover on timer expiry (102), class = Protocol Error
(6) ]
Cause data 0: 38 (56)
Cause data 1: bb (187)
Cause data 2: 5e (94)
-- Processing IE 8 (Cause)
NEW_HANGUP DEBUG: Calling q931_hangup,
[Asterisk-Users] TE410P - Dialogic D240SC
Basically, have an old IVR application running under Apex's Omnivox software on a box with 4 old intel dialogic D240SCs, and would like to allow remote clients to gain access to aforementioned IVR application via softphone, 7960, ata, etc via asterisk with a TE410P. Unfortunately, all I know about the dialogics is that they're configured for D4/AMI, which I believe I have configured right in zaptel.conf: span=1,2,0,d4,ami, etc, etc All spans show green with crossover T1s, but no matter what I try, I get Unable to create channel of type 'Zap' whenever I try to dial out of the TE410P. I've tried various signalling types in zaptel.conf and zapata.conf (pri_cpe, pri_net, em_w), but to no avail. Also unfortunately, I'm rather newbish to asterisk as well as telco on the whole. Does anyone have any suggestions or tips? The online documentation is.. lacking, to say the least. Thanks in advance, -- Cy ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] H.323 Audio problem UPDATE
Tommy, Still waiting from you whether the CDRs are recorded with cdr_csv. This is working just fine for me. Michael. T. Chan wrote: Hi, Scott. Are you telling me that this native h.323 has been hardcoded with fast start? Can you tell me where in ast_h323.cpp it is that you disabled this faststart? Have you tried using the Stable cvs of the Asterisk. Can you let me know which version of the OH323 are you using ? Is it the 0.6.2A? Which version of the Pwlib and OpenH323 you used, is it the newest version as stated? Did you apply the patch? I tried using this driver, but I have problem with cdr_mysql, it is not recording cdr. Please share your information, thanks alot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Sunday, June 27, 2004 6:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H.323 Audio problem UPDATE Update on this problem: I gave up on the native h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even NetMeeting (for the first time). Notes to others who want to try OH323: * The installation is a bit more complicated than h323. Follow the instructions in the ReadMe file exactly. * You must choose and install the proper versions of PWLib and OpenH323, as stated. * Don't forget to edit the Makefile as stated. Some load testing to following this week, but I'm encouraged! Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: H.323 Audio problem UPDATE
I have (as I have mentioned before) 2 identical servers connected to to
same cisco gatekeeper.
Server 1 works fine with no audio problems, server 2 is using cvs head
and there is no audio when connected.
using same configs on both servers (RH9). Disabling faststart didn't
help me.
I have spent some time plugging in exstra debug statements and comparing
the 2 servers. Here is one thing I find
a bit strange about the the non working server and its easy to
reproduce. I think that my no-working server would be
working if my gatekeeper was supporting GSM which it doesn't so I cannot
verify my claim here.
In h323.conf:
disallow all
allow alaw.
start '*'
h.323 show codecs
Allowed Codecs:
Table:
G.711-ALaw-64k{sw} 1
Set:
0:
0:
G.711-ALaw-64k{sw} 1
which is ok afaik
make 1 call (which passes no audio)
h.323 show codecs
Allowed Codecs:
Table:
(empty now.)
the endPoint-GetCapabilities(); returns me an empty string now.
The only codec that 'survives' is for what ever reason the gsm codec.
I will continue to see if I can pinpoint this issue. (I hope that I am
not of on some wild goose chase).
Freddi
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[Asterisk-Users] Meetme
Hi people, I have a user that forgets to hangup his conference calls, so they go on forever. Is there a way of limiting the duration of a conf call? Thanks in advance, Pablo -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Meetme
Pablo Endres wrote: Hi people, I have a user that forgets to hangup his conference calls, so they go on forever. Is there a way of limiting the duration of a conf call? Thanks in advance, Pablo Try using ABSOLUTETIMEOUT before starting the conference? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Unable to forward voice
On Mon, 2004-06-28 at 05:11, administrator tootai wrote: Unable to find a path from G723 to ALAW Unable to find a path from ULAW to G723 Asked to transmit frame type 4, while native format is 1 (read/write = 8/4) Unable to forward voice Even if I force my sip.conf to use only g723.1 I have the same result. BTW, if I want to modify my codecs in a sip context, it's not taking in account by asterisk. Is'it normal behaviour? Asterisk is trying to convert from ALAW to G723.1. Asterisk can't do that. Don't use G723.1. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Eating Digits
When I call a PBX system and enter digits, Asterisk is eating
away some digits. For example when I call ATT and when the
system prompts me to enter my phone number, Asterisk eats
away some digits, so ATT does not get the number that I
entered. I am using the extensions.conf as it came from the
install with some additions. I added longdistance to the
default context. Please help!
[default]
include = mainmenu
include = longdistance
exten = _9X.,1,Dial(ZAP/1/${EXTEN:1})
Try exten = _9X.,1,Dial(ZAP/1/ww${EXTEN:1})
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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Re: [Asterisk-Users] Protocol Error (6) using Zaphfc
Hi - And thanks for the answer!
Unfortunately i get the exact same result with the g1 instead on just 1.
Kind Regards
NRB
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 1:56 PM
Subject: Re: [Asterisk-Users] Protocol Error (6) using Zaphfc
Hei,
please never try to dial out on a particular b channel, you have to dial
out on a zaptel group which includes both b channels of the BRI line.
In a p2mp setup YOU cannot know which b channel will be chosen!
exten = _X.,1,Dial(ZAP/g1/${EXTEN})
will do(note the 'g')
best regards
Klaus
Am Mo, 2004-06-28 um 12.45 schrieb nrb:
Hi!
Has anybody seen anything like this using zaphfc?
On outgoing calls (via isdn) , the line gets hung-up as soon as the
called
party answers.
As seen below i get some protocol error (6) - but i'm not sure if this
is
related to the hang-up which apparently comes a little earlier?!
Incomming calls on the isdn (zaphfc) interface is working just fine
(P.S. what about the D-channel going up down all the time - is that
normal? )
Kind Regards
NRB
Setup
Bri-stuff - 0.0.20
Asterisk CVS-HEAD-06/23/04-15:45:48 built by
[EMAIL PROTECTED] on a
i686 running Linux
Zapata.conf:
[channels]
switchtype = euroisdn
; p2mp TE mode
signalling = bri_cpe_ptmp
; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; p2p NT mode
;signalling = bri_net
pridialplan=local
prilocaldialplan=local
echocancel=yes
immediate=yes
group = 1
context=demo
channel = 1-2
Zaptel.conf:
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
Example where a sip client (2203) is calling 7024
From Asterisk:
== D-Channel on span 1 down
== D-Channel on span 1 up
-- Executing Dial(SIP/2203-5779, Zap/1/7024) in new stack
-- Making new call for cr 135
Protocol Discriminator: Q.931 (8) len=32
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: SETUP (5)
Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer
capability:
Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Ext: 1 User information layer 1: A-Law (35)
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive
Dchan: 0
ChanSel: B1 channel
]
Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number passed network
screening
(1) '2203' ]
Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '7024' ]
Sending Complete (len= 0)
-- Called 1/7024
Protocol Discriminator: Q.931 (8) len=7
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: CALL PROCEEDING (2)
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive
Dchan: 0
ChanSel: B1 channel
]
-- Processing IE 24 (Channel Identification)
Protocol Discriminator: Q.931 (8) len=12
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: ALERTING (1)
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Network beyond the interworking point (10)
Ext: 1 Progress Description: Inband information or appropriate
pattern now
available. (8) ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Network beyond the interworking point (10)
Ext: 1 Progress Description: Unknown (1) ]
-- Processing IE 30 (Progress Indicator)
-- Processing IE 30 (Progress Indicator)
-- Zap/1-1 is ringing
Protocol Discriminator: Q.931 (8) len=15
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: CONNECT (7)
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Public network serving the remote user (4)
Ext: 1 Progress Description: Unknown (4) ]
Time Date (len= 5) [ 04-06-28 11:58 ]
-- Processing IE 30 (Progress Indicator)
-- Processing IE 41 (Date/Time)
Protocol Discriminator: Q.931 (8) len=4
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: CONNECT ACKNOWLEDGE (15)
-- Zap/1-1 answered SIP/2203-5779
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
Connect
Request
Protocol Discriminator: Q.931 (8) len=8
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: DISCONNECT (69)
Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location:
Private network serving the local user (1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]
-- Hungup 'Zap/1-1'
== Spawn extension (intern, 7024, 1) exited non-zero on
'SIP/2203-5779'
Protocol Discriminator: Q.931 (8) len=4
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: RELEASE (77)
-- Channel 1, span 1 got hangup
[Asterisk-Users] AGI-Exec Problem
Hello,
I am having some trouble with the Asterisk::AGI perl library. It seems
that the AGI-Exec() command is causing me a problem.
Here's the line in my AGI code: $AGI-exec('Record',$vmfile:wav, 30);
I'm trying to record voicemail to the file name stored in $vmfile with
a silence timeout of 30. However, this is not being parse by AGI or
Asterisk correctly, since I get the following output from debug level
5:
-- AGI Script Executing Application: (Record)
Options:(/tmp/asterisk/incident-3893006535:wav,)
-- Playing 'beep' (language 'en')
Jun 22 13:53:06 WARNING[1209214400]: file.c:856 ast_writefile: No such
format 'wav,'
Jun 22 13:53:06 WARNING[1209214400]: app_record.c:221 record_exec:
Could not create file /tmp/asterisk/incident-3893006535
Jun 22 13:53:08 WARNING[1209214400]: file.c:464 ast_openstream: File
/tmp/asterisk/incident-3893006535 does not exist in any format
Jun 22 13:53:08 WARNING[1209214400]: app_agi.c:336 handle_streamfile:
Unable to open /tmp/asterisk/incident-3893006535
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Any ideas on how to make AGI parse this arguement correctly?
Thanks,
Tom Daly
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Re: [Asterisk-Users] Unable to forward voice
Eric Wieling a écrit : On Mon, 2004-06-28 at 05:11, administrator tootai wrote: Unable to find a path from G723 to ALAW Unable to find a path from ULAW to G723 Asked to transmit frame type 4, while native format is 1 (read/write = 8/4) Unable to forward voice Even if I force my sip.conf to use only g723.1 I have the same result. BTW, if I want to modify my codecs in a sip context, it's not taking in account by asterisk. Is'it normal behaviour? Asterisk is trying to convert from ALAW to G723.1. That's what I was guessing ;-) Asterisk can't do that. Don't use G723.1. Third party only accept g723 or g729. No solution (or buy a g729 license)? What's the reason to not convert to g723? Thanks -- daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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-oh323, new version 0.6.3
Hello all, Bugfix release 0.6.3 is now available. Basically, call indications should work ok now. Also, the OH323 channel variables for incoming calls are set properly (they can be used for special authentication purposes). Download: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] AGI-Exec Problem
$AGI-exec('Record',$vmfile:wav 30);
- Original Message -
From: Tom Daly [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 8:05 PM
Subject: [Asterisk-Users] AGI-Exec Problem
Hello,
I am having some trouble with the Asterisk::AGI perl library. It seems
that the AGI-Exec() command is causing me a problem.
Here's the line in my AGI code: $AGI-exec('Record',$vmfile:wav, 30);
I'm trying to record voicemail to the file name stored in $vmfile with
a silence timeout of 30. However, this is not being parse by AGI or
Asterisk correctly, since I get the following output from debug level
5:
-- AGI Script Executing Application: (Record)
Options:(/tmp/asterisk/incident-3893006535:wav,)
-- Playing 'beep' (language 'en')
Jun 22 13:53:06 WARNING[1209214400]: file.c:856 ast_writefile: No such
format 'wav,'
Jun 22 13:53:06 WARNING[1209214400]: app_record.c:221 record_exec:
Could not create file /tmp/asterisk/incident-3893006535
Jun 22 13:53:08 WARNING[1209214400]: file.c:464 ast_openstream: File
/tmp/asterisk/incident-3893006535 does not exist in any format
Jun 22 13:53:08 WARNING[1209214400]: app_agi.c:336 handle_streamfile:
Unable to open /tmp/asterisk/incident-3893006535
== Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Any ideas on how to make AGI parse this arguement correctly?
Thanks,
Tom Daly
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Re: [Asterisk-Users] Unable to forward voice
On Mon, 2004-06-28 at 09:36, administrator tootai wrote: Third party only accept g723 or g729. No solution (or buy a g729 license)? What's the reason to not convert to g723? Because the patent holders of the G723.1 patents do not want to license their technology for a reasonable fee. Here is the licensing pricing info for G723.1 direct from the patent holder's web site: http://www.dspg.com/technology/LicensePricing.html The patent holders for G729 are not exactly nice people, but at least they license their patents for a reasonable fee. That's why you can purchase the G729 codec for Asterisk, but not the G723.1 codec. This is covered over and over and over again in the mailing list archives. --Eric -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Ok so here's one i have already asked but i don't know if anyone saw it
Has anyone managed to get the 'i' extension to work.
I have included within each context the following
exten = i,1,Goto(wrong-number,s,1)
then in
[wrong-number]
exten = s,1,GotoIf($[${EXTEN:0:2} = 43}]?10:2)
exten = s,2,GotoIf($[${EXTEN:0:2} = 62}]?11:99)
exten = s,10,Goto(main-office,${EXTEN},1)
exten = s,11,Goto(remote-office,${EXTEN},1)
exten = s,99,Congestion
Problem is the i does not seem to work at all , any suggestions ( have
searched the WiKi )
Best Regards
Simon
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Re: [Asterisk-Users] H.323 Audio problem UPDATE
Sorry this has nothing to do with your audio issue, but I noticed you were
able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with asterisk-oh323
0.6.2. I get the following errors when trying to compile the oh323 wrapper
for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get audio
working. (yes, I tried disabling FastStart in ast_h323.cpp - no change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad to
say that everything seems to work so far. Not only does audio work, but
even the handshaking is now working in both OpenPhone and even NetMeeting
(for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and OpenH323, as
stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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Re: [Asterisk-Users] (no subject)
On Mon, 2004-06-28 at 09:55, Simon wrote:
Ok so here's one i have already asked but i don't know if anyone saw it
Has anyone managed to get the 'i' extension to work.
I have included within each context the following
exten = i,1,Goto(wrong-number,s,1)
then in
[wrong-number]
exten = s,1,GotoIf($[${EXTEN:0:2} = 43}]?10:2)
Take a quick moment to excersize the brain here and think about what the
${EXTEN} would evaluate when at exten= s. I doubt it is what you wanted
it to be.
exten = s,2,GotoIf($[${EXTEN:0:2} = 62}]?11:99)
exten = s,10,Goto(main-office,${EXTEN},1)
exten = s,11,Goto(remote-office,${EXTEN},1)
exten = s,99,Congestion
Problem is the i does not seem to work at all , any suggestions ( have
searched the WiKi )
Best Regards
Simon
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--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] Problems Compiling and Loading asterisk-oh323 0.6.2
Michael: I tried that version also and got the following errors. I just upgraded to 0.6.3 version and it gave me the exact same errors. Any clues? PWLib and Openh323 build just fine, maybe path got b0rked ? Thanks. This is just a snippet of the hundred of errors that I got: -- snip -- asteriskaudio.cxx:170: syntax error before `::' asteriskaudio.cxx:173: syntax error before `;' asteriskaudio.cxx:177: ANSI C++ forbids declaration `error' with no type asteriskaudio.cxx:177: conflicting types for `int error[0]' asteriskaudio.hxx:59: previous declaration as `int error' asteriskaudio.cxx:177: invalid initializer asteriskaudio.cxx:180: parse error before `for' asteriskaudio.cxx:180: parse error before `;' asteriskaudio.cxx:180: syntax error before `++' asteriskaudio.cxx:182: ANSI C++ forbids declaration `snprintf' with no type asteriskaudio.cxx:182: initializer list being treated as compound expression asteriskaudio.cxx:183: ANSI C++ forbids declaration `recordArray' with no type asteriskaudio.cxx:183: variable-size type declared outside of any function asteriskaudio.cxx:183: invalid initializer asteriskaudio.cxx:184: parse error before `}' asteriskaudio.cxx:187: parse error before `;' asteriskaudio.cxx:187: syntax error before `++' asteriskaudio.cxx:189: ANSI C++ forbids declaration `snprintf' with no type asteriskaudio.cxx:189: redefinition of `int snprintf' asteriskaudio.cxx:182: `int snprintf' previously defined here asteriskaudio.cxx:189: initializer list being treated as compound expression asteriskaudio.cxx:189: multiple initializations given for `snprintf' asteriskaudio.cxx:190: ANSI C++ forbids declaration `playArray' with no type asteriskaudio.cxx:190: variable-size type declared outside of any function asteriskaudio.cxx:190: invalid initializer asteriskaudio.cxx:191: parse error before `}' asteriskaudio.cxx:202: syntax error before `::' asteriskaudio.cxx:217: parse error before `' asteriskaudio.cxx:225: invalid use of undefined type `class PAsteriskSoundChannel' asteriskaudio.hxx:69: forward declaration of `class PAsteriskSoundChannel' asteriskaudio.cxx: In method `BOOL PAsteriskSoundChannel::Open(...)': asteriskaudio.cxx:232: `deviceFd' undeclared (first use this function) asteriskaudio.cxx:235: `os_handle' undeclared (first use this function) asteriskaudio.cxx:236: `mediaFmt' undeclared (first use this function) asteriskaudio.cxx:237: `frameTm' undeclared (first use this function) asteriskaudio.cxx:238: `frameNm' undeclared (first use this function) asteriskaudio.cxx:239: `packetSz' undeclared (first use this function) asteriskaudio.cxx:240: invalid use of undefined type `class PAsteriskSoundChannel' asteriskaudio.hxx:69: forward declaration of `class PAsteriskSoundChannel' asteriskaudio.cxx:225: incomplete `this' defined here asteriskaudio.cxx: At top level: asteriskaudio.cxx:247: invalid use of undefined type `class PAsteriskSoundChannel' asteriskaudio.hxx:69: forward declaration of `class PAsteriskSoundChannel' asteriskaudio.cxx: In method `BOOL PAsteriskSoundChannel::Close()': asteriskaudio.cxx:255: invalid use of undefined type `class PAsteriskSoundChannel' asteriskaudio.hxx:69: forward declaration of `class PAsteriskSoundChannel' asteriskaudio.cxx:247: incomplete `this' defined here asteriskaudio.cxx:256: `PChannel' undeclared (first use this function) asteriskaudio.cxx:256: parse error before `::' asteriskaudio.cxx:258: confused by earlier errors, bailing out make[1]: *** [asteriskaudio.o] Error 1 -- end snip -- On Yaum al-Ithnain 10 Jumaada al-Awal 1425 06:28 am, Michael Manousos wrote: Use the 0.6.2a version. Michael. Brian Wilkins wrote: Hi, I having a problem compiling the wrapper for oh323. I am running Debian, kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the openh323 version I have is 1.13.5. I execute the following commands first before attempting to compile the wrapper: pwlib_1.6.6: make both openh323 1.13.5 ./configure make opt asterisk-oh323 0.6.2 make I also applied the patch that is said that is needed for openh323 1.13.5. And I get the following errors: make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE - I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:660: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_oh323.c:660: initializer element is not constant chan_oh323.c:660: (near initialization for `oh323_ep_list.lock') I have been sucessful before in compiling all packages before. I still have the libraries installed from the wrapper package. I decided to try and download a newer version of openh323 and pwlib, but they did
[Asterisk-Users] Zap X100P oscillation
Has anyone seen this problem before? I have a server with a single X100P card. The audio level is a low, but if I raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo test. Not at a high frequency but with a noise that is best described as a steam engine starting up. It then starts to clip and crackle. If I bring the gain down to Rx=-2.0 and Tx=0.0 or lower then it settles down but it is very very quiet. I have tried the latest CVS Head with echotraining=800 set and also complied with the aggressive echo cancelling, but nothing seems to help. Ideas welcome! Many thanks Peter Whisker This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] H.323 Audio problem UPDATE
Hi Brian-
I think you have to use 0.6.2a not 0.6.2. Also, you might try the new
version from today: 0.6.3.
And just checking, in your Makefile, that you set ASTERISKSRCDIR =
/usr/src/asterisk. (maybe this is a 0.6.2a thing)
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Wilkins
Sent: Monday, June 28, 2004 8:15 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H.323 Audio problem UPDATE
Sorry this has nothing to do with your audio issue, but I noticed you were
able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with asterisk-oh323
0.6.2. I get the following errors when trying to compile the oh323 wrapper
for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get
audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no
change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad
to say that everything seems to work so far. Not only does audio
work, but even the handshaking is now working in both OpenPhone and
even NetMeeting (for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and
OpenH323, as stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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Re: [Asterisk-Users] H.323 Audio problem UPDATE
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
Michael.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you were
able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with asterisk-oh323
0.6.2. I get the following errors when trying to compile the oh323 wrapper
for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get audio
working. (yes, I tried disabling FastStart in ast_h323.cpp - no change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad to
say that everything seems to work so far. Not only does audio work, but
even the handshaking is now working in both OpenPhone and even NetMeeting
(for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and OpenH323, as
stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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RE: [Asterisk-Users] chan_h323 no audio both ways
Sorry, Tom, I missed this message when it came through. It seems this problem is a continuing issue among the asterisk folk. Tell me, what versions of IOS have you tested with, do you have any of the h323 options enable/disabled in the 5300? -g On Fri, 2004-06-18 at 21:09, T. Chan wrote: Hi Glen, I have had the same problem for quite awhile, since around February, all cvs codes that I have tried, and with h323, I have been getting no audio. I am forced to stay with mid-Jan version of the cvs because of this. I tried using ulaw, g729, but same results, I have in a few occasions dropped a few lines here to ask for advice, but no response, may be we could try to exchange some ideas. Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, June 14, 2004 6:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_h323 no audio both ways I've compiled chan_h323 with the latest cvs code, but my calls don't pass audio. The call connects just fine, as there are no errors reported on either side, nor in a traffic examination with ethereal. I've tried the following: voip phone - asterisk - asterisk - voip phone voip phone - asterisk - asterisk zap - asterisk - asterisk zap - asterisk - cisco cisco - asterisk I'm using ulaw on all connections. Any clues, ideas, or directions would be appreciated. Thanks, Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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:Latest Echo changes
Just spoke to someone at telappliant and there not willing to sell the cards in the uk yet as there not ratified to the UK standard. I've just spoke to someone at digium direct and there forfilling backorders at the moment. I've just placed an order at http://store.yahoo.com/asteriskpbx/newitd1pofxo.html. The guy recokens I they should start shipping at the end of the week. Kind Regards, Chris Bond -Original Message- From: Chris Stenton [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 10:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re:Latest Echo changes Yes but telappliant (the uk disti) have yet to get approval for it in the UK. I've just fired of an e-mail to them as they said they should have it by the end of the month. As you say though you can go direct ... Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Flah Operator Panel show iax2 trunk
We use an IAX2 trunk to our remote office and would like for the receptionist to be able to transfer incoming calls from this trunk. but all calls come in as one user, Is there a way to get a breakout on the flash GUI of the incoming calls? Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cheap (US$120 or less) SIP Phones
Todd at Teledynamics (see wiki page mentioned above) has been very responsive to email, and we did not need to sign up as a reseller to purchase the Uniden phones. Great!! I'll give him a call today and see if I can order one...this looks like a really nice phone for the price and given the reviews from other people I'm actually kind of excitedhow do people get new firmware updates? Is there a website? -- Procrastination is the art of keeping up with yesterday. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (no subject)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: 28 June 2004 16:22
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] (no subject)
On Mon, 2004-06-28 at 09:55, Simon wrote:
Ok so here's one i have already asked but i don't know if anyone saw it
Has anyone managed to get the 'i' extension to work.
I have included within each context the following
exten = i,1,Goto(wrong-number,s,1)
then in
[wrong-number]
exten = s,1,GotoIf($[${EXTEN:0:2} = 43}]?10:2)
Take a quick moment to excersize the brain here and think about what the
${EXTEN} would evaluate when at exten= s. I doubt it is what you wanted
it to be.
Simon says
Ok excuse me for being the big thick plank that i am , but it really is the
fact that when an unrecognised extension is dialled it doesn't seem to
register anywhere at the * . if i monitor asterisk -r i do not see the call
hit the box.
Ta
exten = s,2,GotoIf($[${EXTEN:0:2} = 62}]?11:99)
exten = s,10,Goto(main-office,${EXTEN},1)
exten = s,11,Goto(remote-office,${EXTEN},1)
exten = s,99,Congestion
Problem is the i does not seem to work at all , any suggestions ( have
searched the WiKi )
Best Regards
Simon
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[Asterisk-Users] zaptel compile error
have just updated the sources from cvs when i compile zaptel i get following error can help me ? nicolas snip zaptel.c: In function `zt_ctl_ioctl': zaptel.c:3042: warning: assignment from incompatible pointer type zaptel.c:3044: warning: assignment from incompatible pointer type zaptel.c:3052: error: structure has no member named `close' zaptel.c:3053: error: structure has no member named `set_mode' zaptel.c:3054: warning: assignment from incompatible pointer type zaptel.c: In function `__zt_putbuf_chunk': zaptel.c:5517: warning: implicit declaration of function `hdlc_netif_rx' make: *** [zaptel.o] Error 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] H.323 Audio problem UPDATE
Michael:
Yes I did.
On Yaum al-Ithnain 10 Jumaada al-Awal 1425 11:28 am, Michael Manousos wrote:
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
Michael.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you
were able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with
asterisk-oh323 0.6.2. I get the following errors when trying to compile
the oh323 wrapper for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get
audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no
change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad to
say that everything seems to work so far. Not only does audio work, but
even the handshaking is now working in both OpenPhone and even NetMeeting
(for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and OpenH323,
as stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones
On Monday 28 June 2004 15:56, Chris Hirsch wrote: Todd at Teledynamics (see wiki page mentioned above) has been very responsive to email, and we did not need to sign up as a reseller to purchase the Uniden phones. Great!! I'll give him a call today and see if I can order one...this looks like a really nice phone for the price and given the reviews from other people I'm actually kind of excitedhow do people get new firmware updates? Is there a website? FYI - recent changes in chan_sip (RFC3581 support) will cause the UIP200 to stop functioning properly. Uniden has no current plans to support this RFC. We are currently working with them to determine if they will make the phones at least ignore the new 'rport' parameter (RFC3581) and continue to function. I should know more later today - stay tuned. Ryan FYI - your phone will come with a support site logon, where you can download firmware and configuration files. -- .. Ryan Courtnage Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 addon mysql
Tommy, I reverted asterisk-addons to 04/01/2004 and I was able to compile it with the latest asterisk CVS. Your a lifesaver. Ive been pondering over this problem for over a week now. Thanks! -- Harold Workman CCNA, CCNP Cytel Communications [EMAIL PROTECTED] Ph. 281-449-4000 x3098 Quoting T. Chan [EMAIL PROTECTED]: cvs checkout -D mm/dd/yy asterisk-addons -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: Monday, June 28, 2004 1:03 AM To: [EMAIL PROTECTED]; T. Chan Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk addon mysql Tommy, Thanks, how do i get the older version of asterisk-addons? -- Harold Workman Quoting T. Chan [EMAIL PROTECTED]: Hi, I got the same thing, so what I did was for the asterisk-addons, I used CVS April instead of the most current CVS and it worked. Of course, I would have liked to use the most current CVS of asterisk-addons as well, but since the old version works with the most current version of asterisk anyways, I left it like that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: Sunday, June 27, 2004 3:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk addon mysql hi, ive read through the last few posts with people having problems compiling the asterisk-addons for mysql support, and none of them have helped me resolve my compile problem. I currently have -- CVS-06/24/04-22:20:31 and downloaded asterisk-addons. I compiled * first then asterisk-addons, have added CFLAGS+=-I../asterisk/include When I try to make install for asterisk-addons i get [EMAIL PROTECTED] asterisk-addons]# make clean ; make install rm -f *.so *.o .depend cc -fPIC -I../asterisk -D_GNU_SOURCE -I../asterisk/include -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function decla ration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 I have MySQL-server and devel upgraded at version 4.0.20 on a Fedora Core 1. I would really love to have mysql support Harold Workman This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] H.323 Audio problem UPDATE
Ok,
I got it all to work finally. I removed everything and started from
scratch. I also got the latest version of asterisk from the CVS. I built
PWLib, then applied the patch to oh323 1.13.5 then built oh323, and finally
built and installed the wrapper (0.6.3). I just started up Asterisk and
everything is working fine. Thanks for all the help -
On Yaum al-Ithnain 10 Jumaada al-Awal 1425 11:28 am, Michael Manousos wrote:
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
Michael.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you
were able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with
asterisk-oh323 0.6.2. I get the following errors when trying to compile
the oh323 wrapper for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get
audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no
change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad to
say that everything seems to work so far. Not only does audio work, but
even the handshaking is now working in both OpenPhone and even NetMeeting
(for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and OpenH323,
as stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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[Asterisk-Users] Chan_Capi Down
Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI -- data = @89930:0107901723168212 -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/2 == CAPI Call CAPI[contr1/89930]/2 -- CONNECT_CONF ID=003 #0x000d LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 Controller/PLCI/NCCI= 0x Info= 0x2002 -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, h, 1) exited non-zero on 'SIP/ePfd-7515' dmesg shows: isdn_dc2minor: di(0) ch(-1072539760) invalid capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up kcapi: notify up contr 2 capidrv: controller 2 up isdn_dc2minor: di(1) ch(-1072539760) invalid capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled capi: controller 2 up kcapi: notify up contr 3 capidrv: controller 3 up isdn_dc2minor: di(2) ch(-1072539760) invalid capidrv-3: now up (2 B channels) capidrv-3: D2 trace enabled capi: controller 3 up kcapi: notify up contr 4 capidrv: controller 4 up isdn_dc2minor: di(3) ch(-1072539760) invalid capidrv-4: now up (2 B channels) capidrv-4: D2 trace enabled capi: controller 4 up I hope, that you could help me... Thanks Felix Deierlein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vonage and Asterisk integration
All, I have been thru the archives and all the relevant URLs sent to me. I have sent e-mail to those who have gone before me and are attempting to accomplish the same goal no one has it working?. Doesnt anyone have a WORKING asterisk pbx that hooks into vonage? Thanks, Jerry Roy 562-305-9545
[Asterisk-Users] Re: 'a' and 'o' extensions do not work with app_voicemail.c (was: Newbie needs help)
I've been doing some debugging on this and I think it's a code problem.
I'm by no means an expert on Asterisk or how it is written or
implemented, but the following patch to app_voicemail.c fixes the
issue. With this code change, Asterisk correctly transfers to the 'a'
and 'o' extensions as I'd expect them to.
As I said, I'm not an expert, so I would strongly recommend against
committing this as-is... someone please interpret why this works and
fix the root problem (or help me understand why this works so I can fix
the root problem).
Index: app_voicemail.c
===
RCS file: /usr/cvsroot/asterisk/apps/app_voicemail.c,v
retrieving revision 1.119
diff -C3 -r1.119 app_voicemail.c
*** app_voicemail.c 26 Jun 2004 16:06:19 - 1.119
--- app_voicemail.c 28 Jun 2004 17:58:17 -
***
*** 1727,1735
make_dir(dir, sizeof(dir), vmu-context, ext, INBOX);
if (mkdir(dir, 0700) (errno != EEXIST))
ast_log(LOG_WARNING, mkdir '%s' failed: %s\n,
dir, strerror(errno));
! if (ast_exists_extension(chan,
strlen(chan-macrocontext) ? chan-macrocontext : chan-context, o,
1, chan-callerid))
strcat(ecodes, 0);
! if (ast_exists_extension(chan,
strlen(chan-macrocontext) ? chan-macrocontext : chan-context, a,
1, chan-callerid))
strcat(ecodes, *);
/* Play the beginning intro if desired */
if (!ast_strlen_zero(prefile)) {
--- 1727,1735
make_dir(dir, sizeof(dir), vmu-context, ext, INBOX);
if (mkdir(dir, 0700) (errno != EEXIST))
ast_log(LOG_WARNING, mkdir '%s' failed: %s\n,
dir, strerror(errno));
! if (ast_exists_extension(chan, chan-context, o, 1,
chan-callerid))
strcat(ecodes, 0);
! if (ast_exists_extension(chan, chan-context, a, 1,
chan-callerid))
strcat(ecodes, *);
/* Play the beginning intro if desired */
if (!ast_strlen_zero(prefile)) {
***
*** 1768,1775
strncpy(chan-exten, a, sizeof(chan-exten) -
1);
if (!ast_strlen_zero(vmu-exit)) {
strncpy(chan-context, vmu-exit,
sizeof(chan-context) - 1);
- } else if
(!ast_strlen_zero(chan-macrocontext)) {
- strncpy(chan-context,
chan-macrocontext, sizeof(chan-context) - 1);
}
chan-priority = 0;
free_user(vmu);
--- 1768,1773
On Jun 26, 2004, at 8:52 PM, Chad Scott wrote:
I've been banging my head on a brick wall for about an hour now trying
to understand why the following doesn't work (which is even provided
as an example in the distribution!).
The goal is to create a voicemail-only extension not associated with a
phone. I'd rather not have an extension dedicated to VoicemailMain(),
so I would like the user to be able to hit '*' during the introductory
message and be prompted for a password.
For whatever reason, this doesn't work as expected. The first
section, macro-stdexten, is what is provided in the distribution. It
defines exten = a,1,VoicemailMain(${ARG1}), which should match the
return extension from Voicemail() if the user presses '*'. Neither
this nor my vmonly macro do this properly. The '*' key instead does
nothing.
Am I not understanding macros properly?
Thanks,
Chad
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as
well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring
the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS}) ; Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they
press #, return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy,
send to voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If
they press #, return to start
exten = s-.,1,Goto(s-NOANSWER,1) ;
Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If
they press *, send the user into VoicemailMain
[macro-vmonly]
exten = s,1,Voicemail(${ARG1})
exten = s,2,Hangup
exten = a,1,VoicemailMain(${ARG1})
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Re: [Asterisk-Users] AGI-Exec Problem
On Mon, 28 Jun 2004, Tom Daly wrote:
Hello,
I am having some trouble with the Asterisk::AGI perl library. It seems
that the AGI-Exec() command is causing me a problem.
Here's the line in my AGI code: $AGI-exec('Record',$vmfile:wav, 30);
The proper usage would be:
$AGI-exec('Record', $vmfile:wav|30);
I guess it isn't clearly documented in my code/examples so I'll try to add
some in before the next release. When it was implemented the | was the
only seperator in asterisk, it wasn't until many months later that the
(,,,) args were implemented
James
http://asterisk.gnuinter.net
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RE: [Asterisk-Users] CDRs, Conferencing, and MeetMe
Jeff Workman wrote:
O
--On Wednesday, June 23, 2004 4:26 PM -0400 Roger Gulbranson
[EMAIL PROTECTED] wrote:
On Wed, 2004-06-23 at 15:39, Jeff Workman wrote:
We are developing an on-demand teleconferencing solution. We will
be billing per-minute/per-user.
I've successfully gotten Asterisk to write CDR data to a postgres
database, but with the way I've got things setup right now the CDR
does not have the dialed conference number. We need this
information in order to be able to bill.
As teleconferencing is the only application of the Asterisk box, I
have the dialplan setup to immediately launch into the MeetMe
application and prompt the user for conference number/PIN upon
answering. It appears that the MeetMe module isn't interested in
passing the conference number back to Asterisk when the user
disconnects so that Asterisk can include that information in the
CDR.
Any suggestions on how to do this?
Use Read() to collect conference number. Invoke MeetMe() with said
number. Use as well for CDR.
k, this works. However, I'm having difficulty with getting asterisk to
properly handle a user inputting an invalid conference number. My
extensions look like this:
exten = s,1,BackGround(conf-getconfno)
exten = s,2,Read(CONF)
exten = s,3,AbsoluteTimeout(7200)
exten = s,4,MeetMe(${CONF})
exten = s,5,Goto(s,2)
exten = s,6,Hangup()
exten = T,1,Hangup()
s,5 never executes because MeetMe exits non-zero whenever somebody
dials an invalid conference number. How do I work around this?
-J
Well... I have conferences created ann its properties written into
database:
Conference no, conf name, max users, moderator etc.
Also, I do not use exten = s,priority,application... but a proper
extension number or _X, .
Result, every users call to any of the conferences is saved in the CDR.
Hope that helps...
Ta
Senad
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Re: [Asterisk-Users] CDRs, Conferencing, and MeetMe
On Mon, 2004-06-28 at 12:57, Jeff Workman wrote:
O
--On Wednesday, June 23, 2004 4:26 PM -0400 Roger Gulbranson
[EMAIL PROTECTED] wrote:
On Wed, 2004-06-23 at 15:39, Jeff Workman wrote:
We are developing an on-demand teleconferencing solution. We will be
billing per-minute/per-user.
I've successfully gotten Asterisk to write CDR data to a postgres
database, but with the way I've got things setup right now the CDR does
not have the dialed conference number. We need this information in
order to be able to bill.
As teleconferencing is the only application of the Asterisk box, I have
the dialplan setup to immediately launch into the MeetMe application
and prompt the user for conference number/PIN upon answering. It
appears that the MeetMe module isn't interested in passing the
conference number back to Asterisk when the user disconnects so that
Asterisk can include that information in the CDR.
Any suggestions on how to do this?
Use Read() to collect conference number. Invoke MeetMe() with said
number. Use as well for CDR.
k, this works. However, I'm having difficulty with getting asterisk to
properly handle a user inputting an invalid conference number. My
extensions look like this:
exten = s,1,BackGround(conf-getconfno)
exten = s,2,Read(CONF)
exten = s,3,AbsoluteTimeout(7200)
exten = s,4,MeetMe(${CONF})
exten = s,5,Goto(s,2)
exten = s,6,Hangup()
exten = T,1,Hangup()
s,5 never executes because MeetMe exits non-zero whenever somebody dials an
invalid conference number. How do I work around this?
My first temptation is to create an app called MeetMeExists to check for
the existence of a conference.
It would be pretty easy to hack the MeetMeCount code to do this.
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[Asterisk-Users] Queue hold time in seconds
I'm going to modify the queue announcements to allow for rounded seconds (e.g. we want to know to the tens of seconds. E.g. Average wait 1 minute 20 seconds). I'm going to add the optional announce of seconds to the queue config and a rounding factor (e.g. 10 in our case). The following parameters will be added Queue-announce-seconds (default is off) Queue-seconds (default will be an as yet unrecorded "queue-seconds") Queue-rounding-seconds (default will be 10) Have I missed anything? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendata.co.uk
RE: [Asterisk-Users] CDRs, Conferencing, and MeetMe
Roger Gulbranson wrote:
On Mon, 2004-06-28 at 12:57, Jeff Workman wrote:
O
--On Wednesday, June 23, 2004 4:26 PM -0400 Roger Gulbranson
[EMAIL PROTECTED] wrote:
On Wed, 2004-06-23 at 15:39, Jeff Workman wrote:
We are developing an on-demand teleconferencing solution. We will
be billing per-minute/per-user.
I've successfully gotten Asterisk to write CDR data to a postgres
database, but with the way I've got things setup right now the CDR
does not have the dialed conference number. We need this
information in order to be able to bill.
As teleconferencing is the only application of the Asterisk box, I
have the dialplan setup to immediately launch into the MeetMe
application and prompt the user for conference number/PIN upon
answering. It appears that the MeetMe module isn't interested in
passing the conference number back to Asterisk when the user
disconnects so that Asterisk can include that information in the
CDR.
Any suggestions on how to do this?
Use Read() to collect conference number. Invoke MeetMe() with said
number. Use as well for CDR.
k, this works. However, I'm having difficulty with getting asterisk
to properly handle a user inputting an invalid conference number.
My extensions look like this:
exten = s,1,BackGround(conf-getconfno)
exten = s,2,Read(CONF)
exten = s,3,AbsoluteTimeout(7200)
exten = s,4,MeetMe(${CONF})
exten = s,5,Goto(s,2)
exten = s,6,Hangup()
exten = T,1,Hangup()
s,5 never executes because MeetMe exits non-zero whenever somebody
dials an invalid conference number. How do I work around this?
My first temptation is to create an app called MeetMeExists to check
for the existence of a conference.
Do you mean if conference is active or is it created in meetme.conf?
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[Asterisk-Users] Would this work?
Title: Message
I am trying to
implement a rollover of extensions.
exten = 3000,1,GotoIf($[${line1} =
Congestion]?3:2)exten = 3000,2,Dial(${line1},15,rt)exten =
3000,3,GotoIf($[${line2} = Congestion]?5:4)exten =
3000,4,Dial(${line2},15,rt)exten = 3000,5,GotoIf($[${line3} =
Congestion]?7:6)exten = 3000,6,Dial(${line3},15,rt)exten =
3000,7,GotoIf($[${line4} = Congestion]?1:8)exten =
3000,8,Dial(${line4},15,rt)exten = 3000,9,Hangup
The $line[x]
represents a Zap Channel.
Thanks,
-gcc
Re: [Asterisk-Users] Asterisk Flah Operator Panel show iax2 trunk
Thank you for the prompt reply but when I add 7;8;9, in my button number field the iax2 button goes away. i just got .10 today . On Mon, 2004-06-28 at 11:51, Nicolas Gudino wrote: Hi Justin, Justin Carlson wrote: We use an IAX2 trunk to our remote office and would like for the receptionist to be able to transfer incoming calls from this trunk. but all calls come in as one user, Is there a way to get a breakout on the flash GUI of the incoming calls? I'm working exactly on it right now. The way I am handling the IAX or any other VOIP trunk is maybe limited, but I couldn't find a better aproach. Basically, you can have one line in op_buttons.cfg for IAX users, like IAX2[guest] for Iaxtel. In the button number, you can add as many as you like, eg: 1;2;3;4;5;6. The server then populates the buttons as they are being used. If you have only one call, it will show it in button 1, if you have more, it will use the remaining buttons. If you exceed the number of buttons, the rest of the calls will not show up. This is working now, but only for showing info (in the online demo there are three iaxtel buttons, you can call 17005011506 to see it working). I have to work now on transfers and hangups. If time permits I will finish later today or maybe tomorrow. For anyone interested in Flash Operator Panel, there is a mailing list to discuss about it. You can subscribe sending a mail to [EMAIL PROTECTED] Best regards, ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] H.323 Audio problem UPDATE
Michael Manousos wrote: The performance of the oh323 channel driver is limited by OpenH323. asterisk-oh323 uses the (more complete) RTP implementation offered by the library, and not that of Asterisk. Of course there are pros (adaptive jitter buffer, RTCP implementation) and cons (lower performance). It's up to the user to select the one that performs better for his application. flamePut the crack pipe down./flame We have gone over this before, asterisk-oh323 is limited by the method you implemented to buffer the audio around. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Flah Operator Panel show iax2 trunk
On Mon, 2004-06-28 at 16:02, Justin Carlson wrote: Thank you for the prompt reply but when I add 7;8;9, in my button number field the iax2 button goes away. i just got .10 today . That feature will be available in 0.11, is not complete yet (I'm working on it). Please subscribe to the operator panel mailing list to continue this thread. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weird 7940 issue
Hi all, On my 7940 phone when I dial out I press 9, then the number. After I press the second number (IE: 9,1) the dialtone stops playing just like it should. This is normal and similar to a regular phone. On two of my 7940s the phones continue the dialtone. No matter how many numbers you dial the dialtone does not stop until you press dial. Also, on these two phones a little X appears next to both line appearances after rebooting. They go away an unkown amount of time later. Sometimes only one will go away, sometimes one will stay. I don't know if this is just a delay in registration or if there is a problem. All of my phones are behind NAT. The Asterisk server is not behind NAT. I am running CVS head from the 19th. TIA, -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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-oh323, new version 0.6.3
On Mon, 2004-06-28 at 07:45, Michael Manousos wrote: Hello all, Bugfix release 0.6.3 is now available. Basically, call indications should work ok now. Also, the OH323 channel variables for incoming calls are set properly (they can be used for special authentication purposes). Download: http://www.inaccessnetworks.com/projects/asterisk-oh323 Will it work as a H323 gatekeeper? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Security Vulnerability in Asterisk
This was fixed in cvs HEAD and stable on 4/13/2004 and a new source release was made at the time (version 0.9.0) I'm not sure why it would be brought up on a recent newsletter, it was discussed in here (or maybe on -dev) sometime around 4/15/2004 James On Mon, 28 Jun 2004, Jim Rosenberg wrote: The following is pasted from SecurityFocus Newsletter #254: - Asterisk PBX Multiple Logging Format String Vulnerabilities BugTraq ID: 10569 Remote: Yes Date Published: Jun 18 2004 Relevant URL: http://www.securityfocus.com/bid/10569 Summary: It is reported that Asterisk is susceptible to format string vulnerabilities in its logging functions. An attacker may use these vulnerabilities to corrupt memory, and read or write arbitrary memory. Remote code execution is likely possible. Due to the nature of these vulnerabilities, there may exist many different avenues of attack. Anything that can potentially call the logging functions with user-supplied data is vulnerable. Versions 0.7.0 through to 0.7.2 are reported vulnerable. - What is the status of CVS-current with respect to this? I don't remember seeing any discussion of this issue here; apologies if I missed it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Context for Incomingmsn
Hi List! I use Asterisk as a pure voicemailbox at a customers place. Right now, a telephone uses up two msns, one for the telephone itself, and one for the telephones mailbox. If the user is absent, a telephonecall is redirected to the voicemail msn of that users telephone. The Problem is: The PBX supports a too small number of msns, so I can't give every user a voicemailbox. Mailboxes are assigned after different contexts (in capi.conf the msn option). It would be extremely cool, to create new contexts after incomingmsn, I would only use up 1 msn for all voicemailboxes, and call the context according to the telephone of the user, that was called in his/her absence. Now, that does not work though. The incomingmsn apparently doesn't create a new context. Or does it? Is there a way to do that? Here a diagram of what I want to do: msn 12: the telephone of the user msn 50: the isdn-card of asterisk External call --- 12 --- user present -- phone rings | | v user absent | | v redirection to 50, incomingmsn=12 | | v voicemailbox of 12 is called Thanks in advance, Henning -- There is no normal life [...] there is just life [...] Kevin Jarre, Tombstone |\ _,,,---,,_ ZZZzz /,`.-'`'-. ;-;;,_ |,4- ) )-,_. ,\ ( `'-' '---''(_/--' `-'\_) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Can one send CLID NAME over PRI?
I ran a PRI DEBUG SPAN 1 on our office system. I could not see any FACILITIES messages on outgoing calls over the PRI. So I suppose * does not send the CNAME messages at all on outgoing calls. CLID NAME is just a subset of the generic user to user messaging on ISDN networks. It should be possible to send characterset IE5 u2u messages and have them show up on other ISDN compatible phones. btw. The GSM standard for cellphones is based on ISDN. Any comments on that ? - Alfred -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Sunday, June 27, 2004 8:05 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can one send CLID NAME over PRI? At 2:17 PM -0700 on 6/25/04, Ehud Gavron wrote: Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google... with no success. Ehud [snip] No, CNAM (Caller NAMe) data is looked up via caller ID digit information by the far end (or somewhere in the middle.) This is a vestige of telephone companies wanting to control the entire process from the center, versus the next-generation systems like VoIP wanting to control it from the edges. Welcome to the first ring of hell; it gets warmer the closer you get to SS7. You cannot (yet) transmit this information to any providers in North America, so far as I am aware. You would have to pay your upstream PSTN (SS7) carrier to insert these records in some central database, probably Verisign or the ILEC from which the number is purchased. If you don't own the numbers that you're transiting, then you're out of luck. An interesting trick between Asterisk servers might be to use the UUI (User-to-User Information) data that is part of the PRI q.931 specifications. It is unknown at this time how many carriers actually pass UUI data from end-to-end, and there currently exist only a few sparse patches for Asterisk that deal it. For some additional discussion, use Google or start here: http://lists.digium.com/pipermail/asterisk-dev/2003-September/001748.html JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 Festival, not a happy couple
Hello, I'm in the process of trying to get Festival to work with Asterisk. I followed the install process at http://www.voip-info.org/wiki-Asterisk+festival+installation. To get the Festival to compile I had to add the patch described in the comments. Once added, Festival and the Speech tools compiled without error. How ever, when ever I try to call the test extension, I get a busy signal and the following message in the console: dotlnx*CLI -- Executing Answer(SIP/10020020-55c0, ) in new stack -- Executing Festival(SIP/10020020-55c0, mary had a little lamb) in new stack == Parsing '/etc/asterisk/festival.conf': == Parsing '/etc/asterisk/festival.conf': Found Jun 28 16:35:17 WARNING[360466]: app_festival.c:439 festival_exec: Festival returned ER == Spawn extension (toto-start, 8400, 2) exited non-zero on 'SIP/10020020-55c0' dotlnx*CLI In the console from which I started the Festival server, I sometimes get the following being displayed: SIOD ERROR: unbound variable : tts_textasterisk I've confirmed the festival.conf file. I've confirmed the festival server is running. Asterisk version: v1-0_stable Festival version:1.4.3 Speech tools version: 1.2.3 I've googled but had no luck. Any help/pointers to info would be appreciated... David ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Softphone
Hi, What is the best SIP softphone to use with Asterisk? I have a hard time finding OpenSource SIP soft phone. Regards Arve5 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Zap X100P oscillation
I wonder if your issue and mine are related somehow. I have a asterisk server with 4 FXO cards in it, and when a call comes in one ZAP channel, then dials out another, I hear what could be described as a steam engine starting up. It starts off kinda slower/ quiet, then quickly (in about 2-4 seconds) completely over powers the line. The only way I could stop it was by adjusting the gains. rxgain=-8.5 txgain=4 Seemed to do the trick. As did: rxgain=-6.5 txgain=1 An rxgain of even -8.0 or -6.0 in either case would result in this steam engine sound. -8.5 or -6.5 would make it go away completely. I'm using a CVS checkout from yesterday, and I tried with both echotraining=800 and turning echo cancellation off completely. Neither made any difference. It would be really nice to be able to use a positive rxgain value. I haven't tried with the echo app, but using just one FXO card works fine with almost any rx/txgain value. As soon as the call utilizes two FXO card at the same time, the steam engine sound occurs. On Mon, 2004-06-28 at 16:26 +0100, Whisker, Peter wrote: Has anyone seen this problem before? I have a server with a single X100P card. The audio level is a low, but if I raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo test. Not at a high frequency but with a noise that is best described as a steam engine starting up. It then starts to clip and crackle. If I bring the gain down to Rx=-2.0 and Tx=0.0 or lower then it settles down but it is very very quiet. I have tried the latest CVS Head with echotraining=800 set and also complied with the aggressive echo cancelling, but nothing seems to help. Ideas welcome! Many thanks Peter Whisker This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Softphone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Arve, I've been using kphone (http://www.wirlab.net/kphone/) with success. It's simple and works fine :-) []'s Arve Rasmussen wrote: | Hi, | | What is the best SIP softphone to use with Asterisk? | | I have a hard time finding OpenSource SIP soft phone. | | Regards | | Arve5 | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA4Iq4iLK8unYgEMQRAmDpAJ49AAIqNUN5t1uhvPL0dwt/bub8PgCeOkZn QhjZWvivXvlwdYCO+mz0tWE= =Kslk -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Zap X100P oscillation
Try recompiling your zaptel package without the aggressive echo cancellation enabled. I have aggressive cancellation help before, I but I have also seen it hurt things before. Brian On Mon, 28 Jun 2004 16:26:32 +0100, Whisker, Peter [EMAIL PROTECTED] wrote: I have tried the latest CVS Head with echotraining=800 set and also complied with the aggressive echo cancelling, but nothing seems to help. Ideas welcome! Many thanks Peter Whisker This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Chan_Capi Down
I am also having the same problem. Latest CVS Latest Capi When it does work and you pick up the phone, CAPI disconnects the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ePyron Felix Deierlein Sent: 28 June 2004 18:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Chan_Capi Down Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI -- data = @89930:0107901723168212 -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/2 == CAPI Call CAPI[contr1/89930]/2 -- CONNECT_CONF ID=003 #0x000d LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 Controller/PLCI/NCCI= 0x Info= 0x2002 -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, h, 1) exited non-zero on 'SIP/ePfd-7515' dmesg shows: isdn_dc2minor: di(0) ch(-1072539760) invalid capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up kcapi: notify up contr 2 capidrv: controller 2 up isdn_dc2minor: di(1) ch(-1072539760) invalid capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled capi: controller 2 up kcapi: notify up contr 3 capidrv: controller 3 up isdn_dc2minor: di(2) ch(-1072539760) invalid capidrv-3: now up (2 B channels) capidrv-3: D2 trace enabled capi: controller 3 up kcapi: notify up contr 4 capidrv: controller 4 up isdn_dc2minor: di(3) ch(-1072539760) invalid capidrv-4: now up (2 B channels) capidrv-4: D2 trace enabled capi: controller 4 up I hope, that you could help me... Thanks Felix Deierlein ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Would this work?
MessageIf I am understanding your dialplan snippet correctly, you simply
want * to call extensions in a linear (or even round robin) fashion, ringing
the first one that's not busy correct? This functionality is built directly
into * and needs no special dialplan to implement. Please check the Wiki or
This list about Grouping Zap channels...
- Original Message -
From: AstGrp
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 12:01 PM
Subject: [Asterisk-Users] Would this work?
I am trying to implement a rollover of extensions.
exten = 3000,1,GotoIf($[${line1} = Congestion]?3:2)
exten = 3000,2,Dial(${line1},15,rt)
exten = 3000,3,GotoIf($[${line2} = Congestion]?5:4)
exten = 3000,4,Dial(${line2},15,rt)
exten = 3000,5,GotoIf($[${line3} = Congestion]?7:6)
exten = 3000,6,Dial(${line3},15,rt)
exten = 3000,7,GotoIf($[${line4} = Congestion]?1:8)
exten = 3000,8,Dial(${line4},15,rt)
exten = 3000,9,Hangup
The $line[x] represents a Zap Channel.
Thanks,
-gcc
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[Asterisk-Users] RE: Chan_Capi Down
Same here :-( asterisk show's this error in the same moment i'm trying to pick up an incoming call: Jun 23 13:14:03 ERROR[-1284076624]: chan_capi.c:881 capi_write: dont know how to write subclass 64 This problem starts with cvs update -D 6/21/04 21:00:00 CET If i revert back to cvs update -D 6/21/04 18:00:00 CET the problem is gone. -- original message -- I am also having the same problem. Latest CVS Latest Capi When it does work and you pick up the phone, CAPI disconnects the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ePyron Felix Deierlein Sent: 28 June 2004 18:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Chan_Capi Down Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI -- data = @89930:0107901723168212 -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/2 == CAPI Call CAPI[contr1/89930]/2 -- CONNECT_CONF ID=003 #0x000d LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 Controller/PLCI/NCCI= 0x Info= 0x2002 -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, h, 1) exited non-zero on 'SIP/ePfd-7515' dmesg shows: isdn_dc2minor: di(0) ch(-1072539760) invalid capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up kcapi: notify up contr 2 capidrv: controller 2 up isdn_dc2minor: di(1) ch(-1072539760) invalid capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled capi: controller 2 up kcapi: notify up contr 3 capidrv: controller 3 up isdn_dc2minor: di(2) ch(-1072539760) invalid capidrv-3: now up (2 B channels) capidrv-3: D2 trace enabled capi: controller 3 up kcapi: notify up contr 4 capidrv: controller 4 up isdn_dc2minor: di(3) ch(-1072539760) invalid capidrv-4: now up (2 B channels) capidrv-4: D2 trace enabled capi: controller 4 up I hope, that you could help me... Thanks Felix Deierlein _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Suggestions for Outbound Proxies?
Although nat=yes/qualify=yes can handle some NAT routers, it does not handle all situation in both directions. Our experience suggests that nothing short of a full SIP Outbound Proxy is going to handle things properly. We have tried out ABP International's NATpass and SNOM's NATfilter, both with results that were underwhelming. Has anyone out there tried out a software SIP Outbound Proxy that works? George Pajari netVOICE communications ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] T100P Newbie -- How to test ISDN on DMS100
Murray Hooper wrote: I am trying to work with zap and libpri to do some ISDN circuit testing with Digium T100P. I am trying pritest but can't figure out what dchannel number should be be when I try 24 or 1, I get failed to open dchannel '24'. D-channel is on time slot 24 on our circuit, but don't see what the nomenclature should be to get this test code to run. You might have better luck trying channel 0 or 23, as Asterisk begins counting at 0, not 1. HTH, Trevor ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 VoIP deployment.
Hi, We are looking at deploying Asterisk for about 60 phones. Since we are in a public building, and are a mixed university and federal unit, we must have our phones up near 100% of the time. Currently we have ~60 POTs lines. I am working on moving us to DIDs with a single PRI feeding us. The reason this came up, is that we are planning on growing to over 150 users within the next 2-3 years. My idea is to have a well planned/tested Asterisk server, with a spare on hand (identical, including the digium quad T1 card). We are looking at HP 2650-PWR switches (one of each floor, each with ~30 VoIP phones). For UPS power we are looking at an APC 2200XL-NET with a large extra battery (the big APC one), for each switch, and one set for the server. Our data switch infrastructure would also be on the 2200XL-NET, but we are using HP 2848 Gigabit switches, and the power draw is much less than the 2650-PWR. Two areas that I am running into trouble with is either some FXS/VoIP gateways or a channel bank, for 9 analog devices (6 on the first floor, 3 on the third floor), 7 of which are fax machines. We can probably put a channel bank in on the third floor, and run the analog devices on both floors). Phones. We are looking at either a mix of the Uniden 200 and the Zip 4x4, or all Zip, or all Uniden. I have looked at others but the Snom 205 is not much cheaper than the Zip 4x4. We need 802.3af PoE support. Multi line would also be quite useful. Does anyone know if the problem of not hearing dialed digits during a call still exist with the Uniden. I can probably get enough budget to do all Zip 4x4 phones, but I want to know real world experiences with the two, before I push it one way or the other. Before we make a full decision, I am going to bring one of each in house. Any other suggestions or recommendations would be handy. Both myself, and our systems programmer (him more than me) have worked with Asterisk before, but not for such a deployment. Harry -- Harry McGregor, Computing Manager Tucson Center Support Group - U.S. Geological Survey University of Arizona - Environment and Natural Resource Building 520-670-5574 (office) - [EMAIL PROTECTED] 520-661-7875 (Cell) - [EMAIL PROTECTED] The opinions/statements expressed herein are my own and should not be taken as a position, opinion, or endorsement of the University of Arizona or the U.S. Geological Survey. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modems behind Asterisk - how?
Title: Modems behind Asterisk - how? The configuration I'm building replaces an existing PBX with Asterisk. There are 8 existing modems that people use to call in from the outside to connect to PC(s) on the inside to transfer data, etc. Callers access these modems by calling the main number and then dialing an extension for the modem they want to talk to. What are my options for supporting these modems with Asterisk? Here are two ideas and some pro's and con's: 1. Use 4 Sipuras (approx. $400). Only problem is, I can't get this to work! Sipura says use the G711 codec but it's not working for me. Anybody have this working? 2. Use 8 FXS ports (approx. $700). Haven't tried this yet but it is more expensive. Other suggestions? Advice? Thanks!
RE: [Asterisk-Users] RE: Chan_Capi Down
Thanks I will give that a try. Looks like this may need a bug report? We are all getting the same errors. Outgoing is fine for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Anderson Sent: 28 June 2004 23:26 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: Chan_Capi Down Same here :-( asterisk show's this error in the same moment i'm trying to pick up an incoming call: Jun 23 13:14:03 ERROR[-1284076624]: chan_capi.c:881 capi_write: dont know how to write subclass 64 This problem starts with cvs update -D 6/21/04 21:00:00 CET If i revert back to cvs update -D 6/21/04 18:00:00 CET the problem is gone. -- original message -- I am also having the same problem. Latest CVS Latest Capi When it does work and you pick up the phone, CAPI disconnects the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ePyron Felix Deierlein Sent: 28 June 2004 18:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Chan_Capi Down Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI -- data = @89930:0107901723168212 -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/2 == CAPI Call CAPI[contr1/89930]/2 -- CONNECT_CONF ID=003 #0x000d LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 Controller/PLCI/NCCI= 0x Info= 0x2002 -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, h, 1) exited non-zero on 'SIP/ePfd-7515' dmesg shows: isdn_dc2minor: di(0) ch(-1072539760) invalid capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up kcapi: notify up contr 2 capidrv: controller 2 up isdn_dc2minor: di(1) ch(-1072539760) invalid capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled capi: controller 2 up kcapi: notify up contr 3 capidrv: controller 3 up isdn_dc2minor: di(2) ch(-1072539760) invalid capidrv-3: now up (2 B channels) capidrv-3: D2 trace enabled capi: controller 3 up kcapi: notify up contr 4 capidrv: controller 4 up isdn_dc2minor: di(3) ch(-1072539760) invalid capidrv-4: now up (2 B channels) capidrv-4: D2 trace enabled capi: controller 4 up I hope, that you could help me... Thanks Felix Deierlein _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] SIP Softphone
On Mon, 2004-06-28 at 16:16, Rodrigo P. Telles wrote: I've been using kphone (http://www.wirlab.net/kphone/) with success. It's simple and works fine :-) kphone only supports inband DTMF and so will only support DTMF when using ulaw or alaw. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SpanDSP Scrunching incoming faxes
I tested SpanDSP as an internal extension, and it worked like a charm. Now I am trying to receive faxes from a toll-free nufone DID. I am running g.711uLaw in on this line, so no to cause too many problems. However I receive the following errors after the fax is finished receiving: so the fax comes in Executing RxFAX([EMAIL PROTECTED]:4569]/5, /root/testfax9.tif) in new stack then the errors channel.c:1654 ast_set_read_format: Unable to find a path from ULAW to UNKN app_rxfax.c:253 rxfax_exec: Unable to restore read format on '[EMAIL PROTECTED]:4569]/5 channel.c:1621 ast_set_write_format: Unable to find a path from UNKN to ULAW app_rxfax.c:259 rxfax_exec: Unable to restore write format on '[EMAIL PROTECTED]:4569]/5' I googled around and could not find anything pertaining to this problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Hong Kong VOIP Exchange
Webpage still doesn't work. On Mon, 2004-06-28 at 22:22, [EMAIL PROTECTED] wrote: Dear All, The home page already move to the top, you can try again. Cary LEUNG Network Operator Hong Kong VOIP exchange Network Glynn Condez [EMAIL PROTECTED]: What happened to your website. I am trying to open it but its empty. regards - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 27, 2004 5:20 PM Subject: [Asterisk-Users] Hong Kong VOIP Exchange Dear All, I had setup a server to be Hong Kong VOIP Exchange gateway, do you want to join us, you can find the detail at http://www.voiphk.net Thank You. Cary LEUNG Network Operator Hong Kong VOIP Exchange Network ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Enger [EMAIL PROTECTED] Xintegration ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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_dialogic
I'm planning to buy Dialogic licenses for one of my dialogic boards to use with *. I have already that in the drawer and it's boring me to keep it there with no use. Although, I have heard that it doesn't work for dialout and I would like to confirm if it's true... my plan is the following: Definity --- Asterisk w/ Dialogic -- Asterisk w/ Dialogic --- Definity D-ChannelVOIP/IAX D-Channel Since, I don't have VOIP in the Lucent Definity machines, I think it would be perfect integrated with asterisk and my dust cloud dialogic boards. So, I just want to confirm if it would work with the current chan_dialogic. Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 hyperthreading
On Mon, 2004-06-28 at 14:18, mattf wrote: In my experience HT on with SMP kernel does help. Others have stated on this Thanks. I have had good experiences with RH ES and Core 1 and HT. james signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Security Vulnerability in Asterisk
--On Monday, June 28, 2004 7:21 PM +0200 Michael Sandee [EMAIL PROTECTED] wrote: Other than that... if these problems are not being published when fixed... then other distro's do not have a chance to fix it... (think about distro's that use stable code, but haven't updated to 0.9 because of problems) I have to say -- with somewhat less vehemence -- that I'm another user who sure never noticed that the stable release of Asterisk had moved from 0.7.2 to 0.9x. This should have been an important announcement on *SEVERAL* security grounds. As of 0.7.2, the recommend version of channel H323 had some very serious vulnerabilities that the OpenH323 folks had fixed months previously. This is an opportune time to repeat: H.323 uses ASN.1. ASN.1 is fiendishly complex and is a known bad boy in which many security holes have appeared over the years. It would be naive to think there won't be more. As VOIP hits the big-time and Asterisk joins the ranks of some of the other more famous open-source projects, quick response to security vulnerabilities will be expected. It's nice to know in the case of these format string problems that they were in some sense addressed promptly, but we're not all subscribed to the dev list. A vulnerability that is fixed in CVS head but not back-patched to stable *is not fixed* as far as a large percentage of the user base is concerned. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Do people actually answer questions here?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've only been watching this list for the past 2 days. And it seems to be an one way street: - -Tell about your problems and what you would like to do. Usually no answer. I have to admit I'm rather disappointed with Asterisk, information is probably available but very hard to find ; it seems to be limited to a few privileged people for whom their job is setting up VoIP system Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA4MEcXeDVKqIr3GURAqA6AJ9AfxMx1TMENHyibYcPBN/xXjssNgCZAU2y gnzzkE/UOwSC13Hck57v1MQ= =4qCG -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Security Vulnerability in Asterisk
On Mon, 28 Jun 2004, Jim Rosenberg wrote: I have to say -- with somewhat less vehemence -- that I'm another user who sure never noticed that the stable release of Asterisk had moved from 0.7.2 to 0.9x. This should have been an important announcement on *SEVERAL* security grounds. As of 0.7.2, the recommend version of channel H323 had some very serious vulnerabilities that the OpenH323 folks had fixed months previously. It's nice to know in the case of these format string problems that they were in some sense addressed promptly, but we're not all subscribed to the dev list. A vulnerability that is fixed in CVS head but not back-patched to stable *is not fixed* as far as a large percentage of the user base is concerned. It was fixed in CVS head and stable and at the same time 0.9.0 was released. The existance was noted in the ChangeLog as well that comes with asterisk Asterisk 0.9.0 -- Logging fixes (fixes remote DoS) -- Fixes from the bug tracker -- ADPCM Standardization -- Branch to Stable CVS I'm not sure if there was an announcement posted to the lists about the code release, but it was definitely updated on the asterisk.org page and the wiki James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 79XX Ringers chan_sccp
Hello: Does anyone know how to configure any of the Cisco 79XX phones to get custom ringers when using chan_sccp with Asterisk? I've currently got Asterisk's 05-24-04 CVS-HEAD and Zozo's 0.2 release of chan_sccp. I've tried using ringlist.dat, but that appears to only be for the SIP phones... Thanks for any input, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Do people actually answer questions here?
On Mon, 2004-06-28 at 20:08, Jean-Yves Avenard wrote: I've only been watching this list for the past 2 days. And it seems to be an one way street: - -Tell about your problems and what you would like to do. Usually no answer. You will find many of us will ignore messages if they require us much effort to read. This is a high volume list and a few will slip through the cracks on their own. Then there is the just plain stupid questions. I personally have gotten tired of defending myself when I point out how to behave and be noticed by more people who will answer. In your case, the only problem in message composition is that you should not reply to a message when you choose to start a new message. Breaks threading. I have to admit I'm rather disappointed with Asterisk, information is probably available but very hard to find ; it seems to be limited to a few privileged people for whom their job is setting up VoIP system Thats your next problem. You should look up and see if there is an answer. If you did you would have seen how many times we point people to the wiki that is user contributed. Next when you signed up for this list you should have seen another link for the -doc project. It is shaping up to be a dead-tree documentation project. There is nothing stoping you from asking questions. If someone has the knowledge and the interest to answer you they will. Comments like this though are pushing me hard to not care any more and to unsubscribe. That would be one less person answering questions due to whiners. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Security Vulnerability in Asterisk
--On Monday, June 28, 2004 9:16 PM -0400 James Golovich [EMAIL PROTECTED] wrote: It was fixed in CVS head and stable and at the same time 0.9.0 was released. The existance was noted in the ChangeLog as well that comes with asterisk Good. But the OpenH323 patches were not back-patched for *months*. I'm not sure if there was an announcement posted to the lists about the code release, but it was definitely updated on the asterisk.org page and the wiki Hmm, I see I wasn't subscribed to announce. Shame on me. Well, hopefully in the future new versions of stable can be announced. I'd like to put forward as a good example what the PostgreSQL folks do. They post a kind of weekly progress report. It includes a digest of important patches, and new releases are announced all over the place. The Sunday Asterisk News posts seem to be filling that role here, and are a good thing, which I applaud. A new release of stable should be something to brag about, yes? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cisco 79XX Ringers chan_sccp
The phone will TFTP the file RINGLIST.XML which wants to look something like: CiscoIPPhoneRingList Ring DisplayNameRing ring/DisplayName FileNameringring.raw/FileName /Ring ... the raw files being in a format described on the Cisco site in some of the application docs, and that it will also TFTP as needed. Chris. Hamilton, Andrew wrote (on Jun 28): Hello: Does anyone know how to configure any of the Cisco 79XX phones to get custom ringers when using chan_sccp with Asterisk? I've currently got Asterisk's 05-24-04 CVS-HEAD and Zozo's 0.2 release of chan_sccp. I've tried using ringlist.dat, but that appears to only be for the SIP phones... Thanks for any input, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Do people actually answer questions here?
On Mon, 2004-06-28 at 20:08, Jean-Yves Avenard wrote: I've only been watching this list for the past 2 days. And it seems to be an one way street: - -Tell about your problems and what you would like to do. Usually no answer. Personally I've gotten tired of answering questions over and over again that could be answered my reading the links on www.digium.com (documentation page) or by reading or searching the archives. I've paid my dues by answering questions on the mailing list and on IRC and by doing my Asterisk related website. So I've taken a break from answering questions. Below is a nice list of links for newbies. Useful Asterisk Docs: http://www.digium.com/index.php?menu=documentation (look at the Unofficial Links) and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cheap (US$120 or less) SIP Phones
On Mon, 2004-06-28 at 11:18, Ryan Courtnage wrote: FYI - recent changes in chan_sip (RFC3581 support) will cause the UIP200 to stop functioning properly. Uniden has no current plans to support this RFC. We are currently working with them to determine if they will make the phones at least ignore the new 'rport' parameter (RFC3581) and continue to function. CVS this evening had an option added called nat=never option for phones like the Uniden. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cheap (US$120 or less) SIP Phones
On Tuesday 29 June 2004 01:57, Eric Wieling wrote: On Mon, 2004-06-28 at 11:18, Ryan Courtnage wrote: FYI - recent changes in chan_sip (RFC3581 support) will cause the UIP200 to stop functioning properly. Uniden has no current plans to support this RFC. We are currently working with them to determine if they will make the phones at least ignore the new 'rport' parameter (RFC3581) and continue to function. CVS this evening had an option added called nat=never option for phones like the Uniden. Yes it did - and this took care of the problem with the UIP200. At the same time, Uniden has voluntarily opened a support incident regarding this issue. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP600 stops to send/receive calls
Hi, I'm testing a Polycom IP600. With firmware version 1.1 the phone reboots at any time. With firmware version 1.2, the first reboot was an endless reboot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls. After a reboot is works fine again. We have a * box with many BT101 and softphones working for months without any problem. I'm missing something? it is a bad config file? or it is a phone bug? Thank You for your time. Jorge Mendoza ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 on 64bit ?
Kevin Walsh wrote: Nicholas Bachmann [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Dr. Rich Murphey [EMAIL PROTECTED] wrote: How do you balance the number of active connections per server? In theory, you could use a load balancer. That's as long as you can share the SIP/IAX registrations between the nodes. I'm not sure if that can be done yet - I haven't looked into it. It can. SIP registration info can be stored in a database; see http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers Sorry - I meant the information relating to registrations that have already been made. Like you get when you type sip show users. The database stores everything about a SIP user in the DB: name, secret, IP, etc. Perhaps that's not necessary anyway; The user should attempt to re-register if the connection is broken, and may find itself connecting to a new server automatically. I think you misunderstand; with a LBR and registrations in a database, the user would never know his * box went down unless he was in the middle of a conversation that had the box in the media path. The SIP phone would never have to reregister until the regular registration timeout. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
