Re: [asterisk-users] Interpreting Asterisk Logs
On Thu, 2008-10-09 at 12:51 +0800, Darren Murphy wrote: Hi, Can anybody point me to an online resource that will assist with interpreting Asterisk log files? I note that a similar question was asked in this forum some time ago (http://lists.digium.com/pipermail/asterisk-users/2007-June/189793.html), which doesn't appear to have received any responses. On that occasion, the OP was seeking a log parser - I'm looking for more of a general reference guide. I'm quite new to Asterisk, and VOIP in general, and I'm struggling to understand what many of logged messages mean. The current approach I am taking is to google for specific messages (or parts thereof) - and this has been somewhat fruitful, if not quite tedious. It would be nice to have a reference guide that lists the most common log messages, and what they mean. Does such a guide exist? thanks, Darren Hi Darren, I also would find such a guide very helpful. Probably something is documented in source code. It would be interesting to know whether the kind people who are working on asterisk documentation project have had thoughts on this aspect. Regards, Roberts ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setup for fax machine
On Sun, 12 Oct 2008, sean darcy wrote: Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten = fax,1,Dial(DAHDI/1) ; the fax machine exten = fax,2,Hangup() exten = s,1,Answer() exten = s,2,Dial(DAHDI/2) ; internal extension . Would this work? I'll need another TDM410 card to do this, so I'd like some reassurance before I go purchase it. You need to Answer() the call first, then insert a Wait(2). During that time, asterisk will be listning for fax tone and jump to the fax extension if it hears them. So: exten = s,1,Answer() exten = s,n,Wait(2) Ringing() Dial(DAHDI/2) Hangup() and exten = fax,1,Dial(DHADI/1) exten = fax,n,Hangup() But at this point why not just feed it into RxFax? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime queue_log to mySQL backport to 1.4
http://ftp.iq-labs.net/queue_log- 1.4/asterisk_queue_log_realtime_1.4.19.patch This uses standardized realtime/mysql library from asterisk addons. For it to support SQL inserts in 1.4, you would also need to apply both patches from (1 for asterisk, another for asterisk-addons) http://ftp.iq-labs.net/realtime_store_destroy-1.4/ This will later allow you to upgrade to 1.6 and having everything working without patching. I have patched in asterisk 1.4 . main/logger.c . include/asterisk/config.h . main/config.c I have patched in asterisk-addons 1.4 . res/res_config_mysql.c I have re-installed asterisk and asterisk-addons. I created a database called db1 and in there created a table called queue_log as per instruction http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL I changed /etc/asterisk/extconfig.conf to add the following line: [settings] queue_log = mysql,db1 I changed /etc/asterisk/res_mysql.conf to add the following: [general] dbhost = localhost dbname = db1 dbuser = user dbpass = password dbport = 3306 dbsock = /var/lib/mysql/mysql.sock 1) However, whenever I perform an agent login, no row is written to table queue_log. I checked /var/log/asterisk/queue_log and a new entry is written there. 2) I set debug to 10 on the console in asterisk and re-did the test but there were no error messages in /var/log/asterisk/messages. 3) I set debug on in mysqld and there are no information for inserting into table queue_log, except the cdr logging as below. Tcp port: 0 Unix socket: (null) Time Id CommandArgument 081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1 2 Connect [EMAIL PROTECTED] on db1 081013 16:00:32 1 Query INSERT INTO cdr_log ... 081013 16:01:42 1 Query INSERT INTO cdr_log ... Is there anyone who can help me? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about echo cancelation
2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED] snip Where did you hear that media gateways filter one-way only? Hi , Reading over this reply, it seems to me that having EC working in one direction is not a so well known fact. Could we say : 1. EC working in one direction is the general case but some of us were not aware of that (I was not, for instance) 2. or this is not so widespread use and some private architectures involve EC working in both directions ? For instance, what about Cisco Call Manager ? Reading this http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080149a1f.shtmland specifically reading IP Phone user hears echo paragraph, I can't say which direction(s) is (are) involved (as article don't mention which voice is echoed). Another interesting thing to note in this paper is the solution is to use a load ID on the IP phone, which includes echo suppression on the handset and headset (PSTN phone user hears echo). Does it imply echo cancelling is mostly done inside phones ? In an IP phone, is it more cost effective to use 2-wires connections (which produces echo) AND echo cancellation systems to suppress echo than to use 4-wires connections which don't produce echo in the first place ? Does it also imply that Cisco IP phones use 2-wires connections ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Mon, Oct 13, 2008 at 09:57:33AM +0800, GNUbie wrote: Hello Tzafrir, On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This means Zaptel gets silence from Asterisk. What codecs are used? What do you see on 'sip show channels'? I am using the following codecs: # asterisk -rx 'sip show settings' | grep Codecs Codecs: 0xe (gsm|ulaw|alaw) Below is the CLI output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-081d11d0, Zap/4/1234567) in new stack -- Called 4/1234567 *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.101.102 102 3c27a6824ba 00101/2 0x4 (ulaw) No Rx: INVITE 1 active SIP channel *CLI core show channels Channel Location State Application(Data) Zap/4-1 [EMAIL PROTECTED] Dialing AppDial((Outgoing Line)) SIP/102-081d11d0 [EMAIL PROTECTED]:1 RingDial(Zap/4/1234567) 2 active channels 1 active call So the call is not established yet, right? This is not a temporary state? Can you call from the FXO to Asterisk? (e.g.: to echo test) There is no problem with an inbound calls. I just tried to call the echo test extension number from my mobile phone via FXO/POTS and it works fine. I can hear my own voice. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about echo cancelation
Just a note that deserves to be reminded is http://lists.digium.com/pipermail/asterisk-dev/2006-January/017774.html If Generally speaking there is only one direction of echo cancellation needed is true, EC works in one way ... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about echo cancelation
This one is also a must-read http://lists.digium.com/pipermail/asterisk-dev/2007-May/027536.html except that is the following scheme, I'm wondering if arrows and RX/TX legends are coherent ... What is written : TX * --- * TDM X Y RX + --- - What I would write : RX * --- * TDM X Y TX + --- - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Tzafrir, On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: So the call is not established yet, right? It is already. The CALLER hears the CALLEE's voice but the CALLEE cannot hear the CALLER's voices. This is not a temporary state? What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Cisco 7906g SIP
When the 'upgrading' process fails, it means that one or more of the required files is missing from the TFTP root folder. Check the logs to see which file it fails on, get that file and you should be good to go. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Mon, Oct 13, 2008 at 9:04 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Tzafrir, On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: So the call is not established yet, right? It is already. The CALLER hears the CALLEE's voice but the CALLEE cannot hear the CALLER's voices. This is not a temporary state? What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you are going to dismiss (the most probable) problem (NAT) without posting configs, I am not sure how much help you will get, you will probably be dismissed as well. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 3:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP When the 'upgrading' process fails, it means that one or more of the required files is missing from the TFTP root folder. Check the logs to see which file it fails on, get that file and you should be good to go. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation
[asterisk-users] Unknown call every 30 minutes on the dot.
Here's some freaky stuff coming from Areski CDR tool: 101. 2008-10-13 03:41:23 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:20 102. 2008-10-13 03:11:30 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 103. 2008-10-13 02:41:23 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 104. 2008-10-13 02:11:22 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 105. 2008-10-13 01:41:21 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 106. 2008-10-13 01:11:21 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 107. 2008-10-13 00:41:29 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 108. 2008-10-13 00:11:21 DAHDI/1... 000 unknown 000 BackGround silence/5 s ANSWERED 00:21 When Asterisk see an incoming call without a caller ID, it sets it to unknown and 000. As you can see from the list above, it happens every 30 minutes almost to the second. It is still happening right now, unless that line is in use, in which case it'll try again 30 minutes later. I did notice this in the /var/log/asterisk/full log: [Oct 13 03:11:30] NOTICE[4243] chan_dahdi.c: Got event 17 (Polarity Reversal)... [Oct 13 03:11:38] WARNING[4243] chan_dahdi.c: CallerID returned with error on channel 'DAHDI/1-1' Normally, it says: [Oct 12 13:23:59] DEBUG[30124] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 1, state 4 [Oct 12 13:23:59] DEBUG[30124] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 1, state 4 [Oct 12 13:23:59] DEBUG[30124] chan_dahdi.c: Polarity Reversal event occured - DEBUG 2: channel 1, state 4, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= = -233440198 Any clues? TDM410P with 2 FXO ports and EC module. Running Fedora Core 9 in init:3 with USB disabled (to prevent IRQ conflicts). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7906g SIP
Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 3:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP When the 'upgrading' process fails, it means that one or more of the required files is missing from the TFTP root folder. Check the logs to see which file it fails on, get that file and you should be good to go. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan --
Re: [asterisk-users] How to enable inbound CLI forX-Lite/Asterisk phone.
Hi,
It is not showing any CLI information even after I have placed that
NoOp(${CALLERID(all)}) function for debugging. Following message was
displayed in debug:
Executing [EMAIL PROTECTED]:3]NoOp(ZAP/6-1,)
What should I do since it is critical to have the callee number.
Please give some more hints.
thanks
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Reynolds
Sent: Friday, October 10, 2008 3:10 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to enable inbound CLI
forX-Lite/Asterisk phone.
Quoting Syed Nasruddin [EMAIL PROTECTED]:
I am using asterisk 1.4.18. I am using it for inbound only call
center.
The SIP phones are X-Lite. Right now when a call is proxied by
Asterisk
to X-Lite the agent only sees asterisk written on its CLI screen. I
want
the agents to be able to view the callees number. Is there any way to
make this happen.
CLI showing as asterisk can indicate absent or withheld number. If
asterisk has it, it should pass it on to X-Lite without any special
settings.
Check to see if asterisk has CLI for the call by putting it in a NoOp in
the dialplan - NoOp(${CALLERID(all)}) would do. Watch asterisk with
verbose set to at least 3.
--
Phil Reynolds
mail: [EMAIL PROTECTED]
Web: http://www.tinsleyviaduct.com/phil/
Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
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[asterisk-users] echo over digital line
Hi, I'm using a 4-port BRI card (b410p) to make and receive calls (via chan_misdn): http://www.digium.com/en/products/digital/b410p.php This card supposedly has hardware echo cancellation. How can I check that echo cancellation is actually ON (taps, etc) on a given misdn channel (like with the zap show channel X command). misdn show channel X doesn't seem to show more info than misdn show channels and in any case there's no info on echo cancellation. I also issued an misdn toggle echocancel mISDN/1-u2528 (on a live call) and flipped several times between echo ON/OFF without subjectively noticing any difference (it's odd because I was calling from a SIP softphone to a remote POTS analog phone and the latter usually experiences echo issues). I couldn't reproduce the echo problem today but in some occasions the remote POTS called party complains they hear echo of their own voice (the called user has a standard analog phone, no headsets, etc., and usually never has echo problems). According to this article: http://www.audiodesignline.com/howto/206800151;jsessionid=UJRLVDJCJT2QMQSNDLPSKHSCJUNN2JVN?pgno=1 I (on the Asterisk system) should take care of the remote user's echo issue, ie. either my BRI card or my Asterisk add-on software should correctly echo cancel. How can one disable hardware echo cancellation altogether on the b410p? Which software echo cancellation can I try on misdn channels (apart from the experimental oslec on misdn)? Is there a way to see the echo cancellation details of a live call? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to enable inbound CLI forX-Lite/Asterisk phone.
Quoting Syed Nasruddin [EMAIL PROTECTED]:
Hi,
It is not showing any CLI information even after I have placed that
NoOp(${CALLERID(all)}) function for debugging. Following message was
displayed in debug:
Executing [EMAIL PROTECTED]:3]NoOp(ZAP/6-1,)
What should I do since it is critical to have the callee number.
First off, is CLI information being presented on that line. If so, you
need to adjust zaptel and asterisk so that they see it - this is a
country-specific matter - what works for one may well not work for
another.
If CLI is not being presented on that line, you need to have that enabled.
--
Phil Reynolds
mail: [EMAIL PROTECTED]
Web: http://www.tinsleyviaduct.com/phil/
Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
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Re: [asterisk-users] One Way Audio Problem
Hello Steve, On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro [EMAIL PROTECTED] wrote: If you are going to dismiss (the most probable) problem (NAT) without posting configs, I am not sure how much help you will get, you will probably be dismissed as well. What particular configs are you looking for? Below is my current setup and scenario: [snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone] SNOM is using the 192.168.101.102 IP address Asterisk is using 192.168.101.1 IP address for its eth1 interface FXO port is connected to the POTS SNOM doesn't need to go out to the Internet in this scenario, AFAIK. Below is my current NAT rules: # iptables -L -v -t nat Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes) pkts bytes target prot opt in out source destination 11460 760K RETURN 0-- anyany 192.168.101.0/24 !192.168.101.0/24 Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination 11408 757K MASQUERADE 0-- anyeth0192.168.101.0/24 anywhere Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination Please advice if you need more information from me. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH Bad
I am running 1.4.10.1 and I am getting garbled MOH from calls within the same LAN with no firewall. Calls sound fine, but every 5-10 seconds the MOH gets garbled. I am using the stock MOH files. Any ideas where/how this could occur? There is no debug showing any issue with MOH. Thanks. Peder ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Mon, Oct 13, 2008 at 10:49 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Steve, On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro [EMAIL PROTECTED] wrote: If you are going to dismiss (the most probable) problem (NAT) without posting configs, I am not sure how much help you will get, you will probably be dismissed as well. What particular configs are you looking for? Below is my current setup and scenario: [snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone] SNOM is using the 192.168.101.102 IP address Asterisk is using 192.168.101.1 IP address for its eth1 interface FXO port is connected to the POTS SNOM doesn't need to go out to the Internet in this scenario, AFAIK. Below is my current NAT rules: # iptables -L -v -t nat Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes) pkts bytes target prot opt in out source destination 11460 760K RETURN 0-- anyany 192.168.101.0/24 !192.168.101.0/24 Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination 11408 757K MASQUERADE 0-- anyeth0192.168.101.0/24 anywhere Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes) pkts bytes target prot opt in out source destination Please advice if you need more information from me. Regards, GNUbie First, drop firewall/iptables/selinux and try again. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ERROR:Failed to create H323 listener
Hi I am trying to get H323 to run on Asterisk, basically I had Asterisk running so I followed this tutorial http://astrecipes.net/index.php?n=286 and got h323 to run on my first server on the second server it is just throwing the error: ERROR:Failed to create H323 listener The whole error is : ERROR: Could not open H.323 listener port on 1720 [Oct 13 09:20:48] ERROR[3608]: chan_h323.c:3166 load_module: Unable to create H323 listener. From /var/log/asteris/h323 09:14:43:478 Error:Bind failed 09:14:43:478 ERROR:Failed to create H323 listener 09:14:43:478 Destroying H323 Endpoint I have checked and nothing is running on 1720, I even tried other ports Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
On Oct 13, 2008, at 9:29 AM, [EMAIL PROTECTED] wrote: IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to enable inboundCLI forX-Lite/Asterisk phone.
How do we adjust zaptel and asterisk for CLI??. Is there some variable
to be set??.. kindly explain keeping in view your country settings this
will give me some hint.thanks
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil
Reynolds
Sent: Monday, October 13, 2008 7:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to enable inboundCLI
forX-Lite/Asterisk phone.
Quoting Syed Nasruddin [EMAIL PROTECTED]:
Hi,
It is not showing any CLI information even after I have placed that
NoOp(${CALLERID(all)}) function for debugging. Following message was
displayed in debug:
Executing [EMAIL PROTECTED]:3]NoOp(ZAP/6-1,)
What should I do since it is critical to have the callee number.
First off, is CLI information being presented on that line. If so, you
need to adjust zaptel and asterisk so that they see it - this is a
country-specific matter - what works for one may well not work for
another.
If CLI is not being presented on that line, you need to have that
enabled.
--
Phil Reynolds
mail: [EMAIL PROTECTED]
Web: http://www.tinsleyviaduct.com/phil/
Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
This message was sent using IMP, the Internet Messaging Program.
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[asterisk-users] Need help for debuging
I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. # thread apply all bt Thread 6 (process 20135): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb7469b80, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb7469b4c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xb22bf9a8, c1=0xa2ae648, config=0xb746a7a0, fo=0xb7469c40, rc=0xb7469c44) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xb22bf9a8, peer=0xa2ae648, config=0xb746a7a0) at res_features.c:1365 #5 0x005a40ba in dial_exec_full (chan=0xb22bf9a8, data=Variable data is not available. ) at app_dial.c:1633 #6 0x005a6a33 in dial_exec (chan=0xfffc, data=0x7fff) at app_dial.c:1680 #7 0x08090bad in pbx_extension_helper (c=0xb22bf9a8, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0xb22bf9a8) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 5 (process 11504): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb51e2e90, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb51e2e5c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xa2db720, c1=0xa12acf8, config=0xb51e37c0, fo=0xb51e2f50, rc=0xb51e2f54) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xa2db720, peer=0xa12acf8, config=0xb51e37c0) at res_features.c:1365 #5 0x004085bb in try_calling (qe=0xb51e3ac0, options=Variable options is not available. ) at app_queue.c:2602 #6 0x0040c6eb in queue_exec (chan=0xa2db720, data=0xb51e8010) at app_queue.c:3344 #7 0x08090bad in pbx_extension_helper (c=0xa2db720, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0xa2db720) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 ---Type return to continue, or q return to quit--- #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 4 (process 24033): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb6c56e90, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb6c56e5c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0x9e9cb80, c1=0xaf0ce2f8, config=0xb6c577c0, fo=0xb6c56f50, rc=0xb6c56f54) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0x9e9cb80, peer=0xaf0ce2f8, config=0xb6c577c0) at res_features.c:1365 #5 0x004085bb in try_calling (qe=0xb6c57ac0, options=Variable options is not available. ) at app_queue.c:2602 #6 0x0040c6eb in queue_exec (chan=0x9e9cb80, data=0xb6c5c010) at app_queue.c:3344 #7 0x08090bad in pbx_extension_helper (c=0x9e9cb80, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0x9e9cb80) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 3 (process 30070): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002f6a9e in __lll_mutex_lock_wait () from /lib/tls/libc.so.6 #2 0x0028800b in _L_mutex_lock_3800 () from /lib/tls/libc.so.6 #3 0x00fa9e27 in pthread_mutex_lock () from /lib/tls/libpthread.so.0 #4 0x080668f5 in ast_read (chan=0x2f6a9e) at channel.c:1945 #5 0x08069d98 in ast_channel_bridge (c0=0xb015fea0, c1=0xa0b5c88, config=0xb4e937a0, fo=0xb4e92c40, rc=0xb4e92c44) at channel.c:3399 #6 0x00548f65 in ast_bridge_call (chan=0xb015fea0, peer=0xa0b5c88, config=0xb4e937a0) at res_features.c:1365 #7 0x005a40ba in dial_exec_full (chan=0xb015fea0, data=Variable data is not available. ) at app_dial.c:1633 #8 0x005a6a33 in dial_exec (chan=0xfffc, data=0x348ff4) at app_dial.c:1680 #9 0x08090bad in pbx_extension_helper (c=0xb015fea0, con=Variable con is not available. ) at pbx.c:574 #10 0x08091e86 in __ast_pbx_run (c=0xb015fea0) at pbx.c:2250 #11 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #12 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #13 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 2 (process 21752): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002f6a9e in __lll_mutex_lock_wait () from /lib/tls/libc.so.6 #2 0x0028800b in _L_mutex_lock_3800 () from /lib/tls/libc.so.6 #3 0x0028bb61 in strcasecmp () from
Re: [asterisk-users] How to enable inboundCLI forX-Lite/Asterisk phone.
Quoting Syed Nasruddin [EMAIL PROTECTED]: How do we adjust zaptel and asterisk for CLI??. Is there some variable to be set??.. kindly explain keeping in view your country settings this will give me some hint.thanks Bearing in mind that these settings are for UK BT (even on other providers here, it is different), in /etc/asterisk/zapata.conf, I enabled the following: distinctiveringaftercid=yes usecallerid=yes cidsignalling=v23 (bell and dtmf are also available - it all depends on what is used in your country and by the provider of your line) sendcalleridafter=2 (this is a UK setting - it may not be needed where you are) I also have callwaitingcallerid=yes, but not sure I have any workable use for it at present. calllerid=asreceived is applied to the FXO port (channel 4 on my card, 6 on yours it seems) As I say, these settings are not necessarily correct for other countries - in particular the signalling and need for sendcalleridafter may be different. Can you connect a caller ID display unit to the line and check that caller ID is presented? If it isn't, you will need to get that enabled too. There is one other problem I have found but it probably won't apply to other countries, so see if this gets you anywhere. -- Phil Reynolds mail: [EMAIL PROTECTED] Web: http://www.tinsleyviaduct.com/phil/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 This message was sent using IMP, the Internet Messaging Program. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP 650 Sidecar
Is the IP 650 sidecar compatible with asterisk? If I pair it with the IP 650 phone, can I have more than 6 lines registered w/ the server? Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: [EMAIL PROTECTED] This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4
Hi John, On Mon, Oct 13, 2008 at 9:51 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: http://ftp.iq-labs.net/queue_log- 1.4/asterisk_queue_log_realtime_1.4.19.patch Haven't you forgotten this one? ;) if you have applied everything correctly - queue_log file shoudln't have any more lines (except init when restarting asterisk). Regards, Atis This uses standardized realtime/mysql library from asterisk addons. For it to support SQL inserts in 1.4, you would also need to apply both patches from (1 for asterisk, another for asterisk-addons) http://ftp.iq-labs.net/realtime_store_destroy-1.4/ This will later allow you to upgrade to 1.6 and having everything working without patching. I have patched in asterisk 1.4 . main/logger.c . include/asterisk/config.h . main/config.c I have patched in asterisk-addons 1.4 . res/res_config_mysql.c I have re-installed asterisk and asterisk-addons. I created a database called db1 and in there created a table called queue_log as per instruction http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL I changed /etc/asterisk/extconfig.conf to add the following line: [settings] queue_log = mysql,db1 I changed /etc/asterisk/res_mysql.conf to add the following: [general] dbhost = localhost dbname = db1 dbuser = user dbpass = password dbport = 3306 dbsock = /var/lib/mysql/mysql.sock 1) However, whenever I perform an agent login, no row is written to table queue_log. I checked /var/log/asterisk/queue_log and a new entry is written there. 2) I set debug to 10 on the console in asterisk and re-did the test but there were no error messages in /var/log/asterisk/messages. 3) I set debug on in mysqld and there are no information for inserting into table queue_log, except the cdr logging as below. Tcp port: 0 Unix socket: (null) Time Id CommandArgument 081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1 2 Connect [EMAIL PROTECTED] on db1 081013 16:00:32 1 Query INSERT INTO cdr_log ... 081013 16:01:42 1 Query INSERT INTO cdr_log ... Is there anyone who can help me? -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP 650 Sidecar
On Mon, Oct 13, 2008 at 10:19 AM, Jeremy Mann [EMAIL PROTECTED] wrote: Is the IP 650 sidecar compatible with asterisk? Yes. Our attendant phone is a 650 with three expansion modules (sidecars). Asterisk can't really tell the difference. The sidecar just gives the phones more buttons for lines or speed dials. If I pair it with the IP 650 phone, can I have more than 6 lines registered w/ the server? The 650 supports up to 12 lines with an expansion module. http://www.polycom.com/usa/en/products/voice/desktop_solutions/soundpoint/desk_phones/soundpoint_ip650.html -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help for debuging
On Monday 13 October 2008 10:29:17 gary wrote: I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. The definition of insanity is doing the same thing over and over again, expecting a different outcome. I told you after your previous post how to find the problem. If you aren't willing to follow those instructions, then there is nobody who can help you. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Norman, On Mon, Oct 13, 2008 at 11:02 PM, Norman Franke [EMAIL PROTECTED] wrote: And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. You mean, canreinvite=no? I already have done line on my sip.conf. Thanks. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Steve, On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro [EMAIL PROTECTED] wrote: First, drop firewall/iptables/selinux and try again. I already turned off the firewall and I don't have SELinux on my system and the problem is still there. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setup for fax machine
On Monday 13 October 2008 01:25:12 am Gordon Henderson wrote: On Sun, 12 Oct 2008, sean darcy wrote: Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten = fax,1,Dial(DAHDI/1) ; the fax machine exten = fax,2,Hangup() exten = s,1,Answer() exten = s,2,Dial(DAHDI/2) ; internal extension . Would this work? I'll need another TDM410 card to do this, so I'd like some reassurance before I go purchase it. You need to Answer() the call first, then insert a Wait(2). During that time, asterisk will be listning for fax tone and jump to the fax extension if it hears them. So: exten = s,1,Answer() exten = s,n,Wait(2) Ringing() Dial(DAHDI/2) Hangup() and exten = fax,1,Dial(DHADI/1) exten = fax,n,Hangup() would you also be able to detect fax tones during the Backgound app? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1 second delay when connecting calls
Hello, Thanks for your replies. We checked our sip.conf and we have canreinvite=no already. I agree it could be a firmware issue. I will get another vendors phone hooked up to the pbx before going crazy with support. Thanks, Neal On Sun, Oct 12, 2008 at 6:14 AM, Vieri [EMAIL PROTECTED] wrote: --- On Sat, 10/11/08, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Try setting canreinvite=no in each of the device sections on a couple of phones, reload chan_sip.so and see if that fixes things. It has fixed the issue when I've tried it. [EMAIL PROTECTED] wrote: Hello, We are using asterisk 1.6, sangoma pri card, and Cisco 7960 phones. When we make or receive calls there is a delay before voice is heard. Anyone have any ideas on where to start to debug or has anyone seen this before. We have played with settings on pri, asterisk, and phones with no change. I'm having the same problem but with ATA-connected analog phones. The ATAs are Grandstream GXW4008 with firmware v. 1.0.1.15. The canreinvite option in sip.conf doesn't change anything for me. Downgrading the GXW4008 solves this issue so this is obviously a firmware bug in my case. I had a vague report once of a user in another installation having this 1-second delay on call connection. That user had a Cisco phone but I don't remember which one. I suggest you check this with Cisco Support if you can. Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
A packet trace will probably show exactly what is happening. Try: tcpdump -nlXs 8192 -i eth0 port 5060 You should be able to see the SIP information going back and forth and will probably show you that your NAT rules are applying when they shouldn't. I agree with first turning off your firewall and testing... but if that actually solves the problem you need to know why. This should tell why. j On Mon, 13 Oct 2008, GNUbie wrote: Hello Norman, On Mon, Oct 13, 2008 at 11:02 PM, Norman Franke [EMAIL PROTECTED] wrote: And reinvite issues in particular. Those have been the only one-way audio problems I've experienced. Setting reinvite=no fixed everything for me. You mean, canreinvite=no? I already have done line on my sip.conf. Thanks. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** Handled by Will's new toy *** ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only worked one way. I ended up rigging * to force the line to be considered answered with a patch. Daniel On Oct 13, 2008, at 8:57 AM, GNUbie wrote: Hello Steve, On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro [EMAIL PROTECTED] wrote: First, drop firewall/iptables/selinux and try again. I already turned off the firewall and I don't have SELinux on my system and the problem is still there. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help for debuging
- Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 11:54 AM Subject: Re: [asterisk-users] Need help for debuging On Monday 13 October 2008 10:29:17 gary wrote: I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. The definition of insanity is doing the same thing over and over again, expecting a different outcome. I told you after your previous post how to find the problem. If you aren't willing to follow those instructions, then there is nobody who can help you. -- Tilghman For some reason, I never received your reply nor my original post. That is why I repost again. Can you repost your reply here? gary ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hi, Am Montag, den 13.10.2008, 10:00 +0800 schrieb GNUbie: Hello Gordon, On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson [EMAIL PROTECTED] wrote: You mention the SIP phone being inside the LAN. Where is the Asterisk box? It is the main gateway of the IP phones and my laptop to the Internet. In this case, the eth1 of the Asterisk box is connected to the LAN and eth0 is connected to the Internet. IME: One-way audio problems are almost always casued by NAT gateways and/or incorrect NAT settings in sip.conf and/or incorrect IP address or extenal proxy settings in the SIP phone. I don't think NAT is involve on this one way audio problem. Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? Regards, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help for debuging
gary schrieb: For some reason, I never received your reply nor my original post. That is why I repost again. Can you repost your reply here? http://lists.digium.com/pipermail/asterisk-users/2008-October/219985.html Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help for debuging
On Monday 13 October 2008 11:19:35 gary wrote: - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 11:54 AM Subject: Re: [asterisk-users] Need help for debuging On Monday 13 October 2008 10:29:17 gary wrote: I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. The definition of insanity is doing the same thing over and over again, expecting a different outcome. I told you after your previous post how to find the problem. If you aren't willing to follow those instructions, then there is nobody who can help you. For some reason, I never received your reply nor my original post. That is why I repost again. Can you repost your reply here? See the archives: http://lists.digium.com/pipermail/asterisk-users/2008-October/219985.html -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote: Eric Chamberlain wrote: Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making outbound calls. In the case of a server like Asterisk, wouldn't sending a register disrupt the flow of inbound calls until the UA that normally handles inbound calls re-registers? Are you using the same credentials as existing extensions to make calls from different extensions? That would seem to be a particularly bad idea. You should be configuring /one/ sip extension per SIP phone. Those extensions that handle outgoing calls only could be put in a different number range, or have a letter prefixed or suffixed to the extension, but you should /not/ be using one configured extension for two different purposes. We're developing the client and don't have control over the server, which may or may not be Asterisk. Adding extra extensions isn't possible. Can OPTION packets be used to verify authentication? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setup for fax machine
On Mon, 13 Oct 2008, Anthony Messina wrote: On Monday 13 October 2008 01:25:12 am Gordon Henderson wrote: On Sun, 12 Oct 2008, sean darcy wrote: Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten = fax,1,Dial(DAHDI/1) ; the fax machine exten = fax,2,Hangup() exten = s,1,Answer() exten = s,2,Dial(DAHDI/2) ; internal extension . Would this work? I'll need another TDM410 card to do this, so I'd like some reassurance before I go purchase it. You need to Answer() the call first, then insert a Wait(2). During that time, asterisk will be listning for fax tone and jump to the fax extension if it hears them. So: exten = s,1,Answer() exten = s,n,Wait(2) Ringing() Dial(DAHDI/2) Hangup() and exten = fax,1,Dial(DHADI/1) exten = fax,n,Hangup() would you also be able to detect fax tones during the Backgound app? I'm really not sure - I think you might have to try it! Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cli commands missing
Hi Eric, Am Sonntag, den 12.10.2008, 18:06 -0700 schrieb Eric Fort: resolve.conf and dns is working. The problem persists. /var/log/asterisk/messages shows a few notices and warnings on res_smdi.c, res_musiconhold.c, and usbradio.c. when I disable loading of these in modules.conf asterisk crashes on load. Eric try to start asterisk in foreground. First stop it and then start it on the shell: asterisk -vv You will see all the messages while autoloading. Some of them might tell You, why a module is not loaded. HTH, Regards, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing CdrTool on free bsd
Hello Friends, Can any one help me installing the CDR TOOL to integrate with freeradius on freebsd? Any help will be highly appreciated. Thanks Mohit DISCLAIMER: The information contained in this message (including any attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and permanently delete this message and any attachments from your system. Any form of dissemination, use, review, distribution, printing or copying of this message in whole or in part is strictly prohibited if you are not the intended recipient of this e-mail. Please note that e-mails are susceptible to change. STARCOMMS PLC shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt or damage to your system. STARCOMMS PLC does not guarantee that the integrity of this communication has been maintained or that this communication is free of viruses, interceptions or interferences. STARCOMMS PLC reserves the right to monitor all e-mail communications, whether related to the business of STARCOMMS or not, through its internal or external networks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
On Mon, Oct 13, 2008 at 12:37 PM, Eric Chamberlain [EMAIL PROTECTED] wrote: We're developing the client and don't have control over the server, which may or may not be Asterisk. Adding extra extensions isn't possible. Can OPTION packets be used to verify authentication? -- Eric Chamberlain Most implementations (including Asterisk) don't challenge OPTIONS, at least I don't think they do... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Hangup
When I sent Hangup using AGI application, Asterisk always return -1: AGI Rx EXEC HANGUP -- AGI Script Executing Application: (HANGUP) Options: ((null)) AGI Tx 200 result=-1 But only send Bye ( Sip message ) when the script is fisnish, not when I send EXEC HANGUP. Why? Thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphone Framework or Libraries
Hi to all, I have a project for Customer Relationship Management interfaced with asterisk, I need send the CallerID to my application (via http or tcp/ip), When the phone rings I need to launch a pop-up windows to the Call Center Agent to display customer info, do you know a framework/libraries to make this, if is possible a softphone embedded into html page for the same function. I need to choice one to suit it to my needs Thanks in advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Text messaging and Asterisk
Can somebody please give a pointer to a complete neophyte (like me) on text messaging, what product can I use to send and automatic text message to a cell phone from within the asterisk dialplan? (the part of the dialplan I have down, the part of the text message no) Thanks C. Savinovich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setup for fax machine
On Mon, Oct 13, 2008 at 12:50 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Mon, 13 Oct 2008, Anthony Messina wrote: On Monday 13 October 2008 01:25:12 am Gordon Henderson wrote: On Sun, 12 Oct 2008, sean darcy wrote: Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten = fax,1,Dial(DAHDI/1) ; the fax machine exten = fax,2,Hangup() exten = s,1,Answer() exten = s,2,Dial(DAHDI/2) ; internal extension . Would this work? I'll need another TDM410 card to do this, so I'd like some reassurance before I go purchase it. You need to Answer() the call first, then insert a Wait(2). During that time, asterisk will be listning for fax tone and jump to the fax extension if it hears them. So: exten = s,1,Answer() exten = s,n,Wait(2) Ringing() Dial(DAHDI/2) Hangup() and exten = fax,1,Dial(DHADI/1) exten = fax,n,Hangup() would you also be able to detect fax tones during the Backgound app? I'm really not sure - I think you might have to try it! Gordon The best way to do this is have a dedicated fax line, do not use fax detect, and route any calls to that zap/dahdi port to where you have the dedicated fax or just keep the fax out of Asterisk entirely. I am used to large scale fax setup, so I am more familiar with DIDs assigned to FAX and then use IAXmodem to Hylafax. Sounds like bigtime overkill for your use. You could look at www.trustfax.com (no affiliation but a happy customer), better service than www.efax.com as far as I am concerned and better pricing. My faxing requirements are very low though. Most customers that insist of dual use POTS lines invariably complain about the delay, which I have to explain, especially if they have caller ID too. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone Framework or Libraries
Tim Panton from Phone From Here was able to implement this functionality when he was at Mexuar so I would check with him. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Melendez Sent: Monday, 13 October 2008 1:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Softphone Framework or Libraries Hi to all, I have a project for Customer Relationship Management interfaced with asterisk, I need send the CallerID to my application (via http or tcp/ip), When the phone rings I need to launch a pop-up windows to the Call Center Agent to display customer info, do you know a framework/libraries to make this, if is possible a softphone embedded into html page for the same function. I need to choice one to suit it to my needs Thanks in advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging and Asterisk
C. Savinovich wrote: Can somebody please give a pointer to a complete neophyte (like me) on text messaging, what product can I use to send and automatic text message to a cell phone from within the asterisk dialplan? (the part of the dialplan I have down, the part of the text message no) IIRC, asterisk currently supports sending text messages only when voice call is already established, so not very usefull yet... PJ Thanks C. Savinovich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ifbyphone/google analytics
Any thoughts? http://www.websitemagazine.com/content/blogs/posts/archive/2008/10/13/go ogle-analytics-track-phone-calls.aspx Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging and Asterisk
I mean is if someone know of an sms server or service that allows me to send outgoing text messaging. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek Sent: Monday, October 13, 2008 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Text messaging and Asterisk C. Savinovich wrote: Can somebody please give a pointer to a complete neophyte (like me) on text messaging, what product can I use to send and automatic text message to a cell phone from within the asterisk dialplan? (the part of the dialplan I have down, the part of the text message no) IIRC, asterisk currently supports sending text messages only when voice call is already established, so not very usefull yet... PJ Thanks C. Savinovich ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone Framework or Libraries
Sorry to hijack this thread...but I'm still looking for a G.722 capable
soft phone other than Eyebeam. I only need a handful of licenses. There
are a couple others available but only to OEMs in 1000 unit quantities.
Any ideas?
Michael
--Original Message Text---
From: Dean Collins
Date: Mon, 13 Oct 2008 13:58:38 -0400
v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}
st1\:*{behavior:url(#default#ieooui) }
Tim Panton from Phone From Here was able to implement this
functionality when he was at Mexuar so I would check with him.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Melendez
Sent: Monday, 13 October 2008 1:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Softphone Framework or Libraries
Hi to all, I have a project for Customer Relationship Management
interfaced with asterisk, I need send the CallerID to my application
(via http or tcp/ip), When the phone rings I need to launch a pop-up
windows to the Call Center Agent to display customer info, do you know
a framework/libraries to make this, if is possible a softphone embedded
into html page for the same function.
I need to choice one to suit it to my needs
Thanks in advance.
Ricardo Melendez
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245
___
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Re: [asterisk-users] Cisco 7906g SIP
Hi, i had the same problem with an 7970g. You should open one of the files, tht end with .loads and compare the files on you tftp-server with the list at the end of this file. Pay attention on case sensitive writing of filenames. I had to changed the jarXYZ.sbn to JarXYZ.sbn an the upgrade was properly done. Greetings Peter David Gibbons schrieb: Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 3:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP When the 'upgrading' process fails, it means that one or more of the required files is missing from the TFTP root folder. Check the logs to see which file it fails on, get that file and you should be good to go. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008
Re: [asterisk-users] ifbyphone/google analytics
On Oct 13, 2008, at 11:21 AM, Dean Collins wrote: Any thoughts? http://www.websitemagazine.com/content/blogs/posts/archive/2008/10/13/google-analytics-track-phone-calls.aspx It wouldn't be to hard to duplicate this with Asterisk. One could export the entire call path through an IVR or call center, time on call, etc. to Analytics. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging and Asterisk
On Oct 13, 2008, at 11:28 AM, C. Savinovich wrote: I mean is if someone know of an sms server or service that allows me to send outgoing text messaging. Are you sending SMS to known users or to any mobile phone user? If you are sending to a fixed user base, track down the email to SMS gateways for their carriers. Then sending an SMS is no different than sending an e-mail. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone Framework or Libraries
Ricardo Melendez wrote: Hi to all, I have a project for Customer Relationship Management interfaced with asterisk, I need send the CallerID to my application (via http or tcp/ip), When the phone rings I need to launch a pop-up windows to the Call Center Agent to display customer info, do you know a framework/libraries to make this, if is possible a softphone embedded into html page for the same function. I need to choice one to suit it to my needs Thanks in advance. Ricardo Melendez Ricardo Try if outcall will do the job you need: http://outcall.sourceforge.net/ Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging and Asterisk
I use the 'generic' file in Postfix to map an email address that is not in use to someone's text messaging address. It'd be [EMAIL PROTECTED] ie: [EMAIL PROTECTED] Then, any email that gets sent to [EMAIL PROTECTED], will get automatically sent to that person's phone. On Mon, Oct 13, 2008 at 3:14 PM, Eric Chamberlain [EMAIL PROTECTED] wrote: On Oct 13, 2008, at 11:28 AM, C. Savinovich wrote: I mean is if someone know of an sms server or service that allows me to send outgoing text messaging. Are you sending SMS to known users or to any mobile phone user? If you are sending to a fixed user base, track down the email to SMS gateways for their carriers. Then sending an SMS is no different than sending an e-mail. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging and Asterisk
I am trying to send text messaging to one caller, maybe about 40 of those per day, whose phone service have expired. All these callers are calling from their cell phones, and I have their caller ids. I will like to send each of them an individual text message (not an email) saying you attempted to use the service, but your service has expired Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Chamberlain Sent: Monday, October 13, 2008 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Text messaging and Asterisk On Oct 13, 2008, at 11:28 AM, C. Savinovich wrote: I mean is if someone know of an sms server or service that allows me to send outgoing text messaging. Are you sending SMS to known users or to any mobile phone user? If you are sending to a fixed user base, track down the email to SMS gateways for their carriers. Then sending an SMS is no different than sending an e-mail. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging and Asterisk
Are you sending SMS to known users or to any mobile phone user? If you are sending to a fixed user base, track down the email to SMS gateways for their carriers. Then sending an SMS is no different than sending an e-mail. If it's for something really important this might not be the best route - here in Canada, with Fido Wireless, the email to sms gateway can be instant, all the way up to 36 hours before you receive them - for something like alerts this is unacceptable. I've called and the response was pretty much too bad so sad - you may be better off with connecting a mobile to the box via Bluetooth or usb, and sending through said mobile.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got event 17 (Polarity Reversal)...
Gordon Henderson wrote: On Sat, 11 Oct 2008, Jorge Mendoza wrote: I founded this behaviour in the past. When the CO provides reversal polarity and the FXO port is configured to ignore polarity events, then a reversal polarity could be detected as ringing if the hardware/software is not well designed or configured. So, if the CO provides polarity reversal, why not set answer and release supervision to yes? We need the flexability to answer either way... Here in the UK the (BT) exchange will do a polarity reversal to signal incoming CLI - it then send the CLI, *then* sends the ring signals, so answering on polarity reversal would be wrong. Answer supervision on reversal polarity applies only to outgoing calls, not incoming ones. They also do a random polarity reversal most nights too - some sort of automated line testing. Eg. from my home box: Oct 7 01:40:28 NOTICE[15581] chan_zap.c: Got event 17 (Polarity Reversal)... Oct 9 03:53:06 NOTICE[19904] chan_zap.c: Got event 17 (Polarity Reversal)... Note the times... Are they just warning alarms or they starts phantom calls? Jorge Gordon Jorge Mendoza Jim Duda wrote: If by default Asterisk ignores all polarity events, then why does it cause the Dialplan to start? I did set answeronplarityswitch to no, however, I have had the problem occur once already, so, you suspicion might be correct. Jim Tzafrir Cohen wrote: On Fri, Oct 10, 2008 at 05:51:29PM -0400, Jim Duda wrote: Tzafrir, Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect this will solve my issue. I never would have know to look for this. Thanks much! You made my day :-) Hmm... I might have misled you. By default Asterisk ignores all polarity events. Using the polarity events can be a useful feature, but I suspect that it is not the cause of your original problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN
Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price competitively with equivalent POTS services and it makes more sense, both in terms of voice quality (4 wire digital to the PABX) and flexibility. Ideally it would allow any combination of two calls, identified by SPID. If anyone has done anything similar, or has any experience with BRI ISDN, I would appreciate input and direction. If anyone knows where documentation exists on configuring ISDN, that information would also be greatly appreciated. Asterisk has a bit of a learning curve, and ISDN BRI isn't the most widely used or covered aspect of it. BTW, I have a strong telecom background, so the theory part of it will not be a problem, only the necessary documentation to apply it to Asterisk. Thanks, Wilton Helm Embedded System Resources ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
I had considered something like this as well, but was convinced to go another direction. I wrote something up about it at the time. http://www.smallnetbuilder.com/content/view/30444/84/ Michael --Original Message Text--- From: Wilton Helm Date: Mon, 13 Oct 2008 14:44:26 -0600 Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price competitively with equivalent POTS services and it makes more sense, both in terms of voice quality (4 wire digital to the PABX) and flexibility. Ideally it would allow any combination of two calls, identified by SPID. If anyone has done anything similar, or has any experience with BRI ISDN, I would appreciate input and direction. If anyone knows where documentation exists on configuring ISDN, that information would also be greatly appreciated. Asterisk has a bit of a learning curve, and ISDN BRI isn't the most widely used or covered aspect of it. BTW, I have a strong telecom background, so the theory part of it will not be a problem, only the necessary documentation to apply it to Asterisk. Thanks, Wilton Helm Embedded System Resources -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got event 17 (Polarity Reversal)...
On Mon, 13 Oct 2008, Jorge Mendoza wrote: Gordon Henderson wrote: On Sat, 11 Oct 2008, Jorge Mendoza wrote: I founded this behaviour in the past. When the CO provides reversal polarity and the FXO port is configured to ignore polarity events, then a reversal polarity could be detected as ringing if the hardware/software is not well designed or configured. So, if the CO provides polarity reversal, why not set answer and release supervision to yes? We need the flexability to answer either way... Here in the UK the (BT) exchange will do a polarity reversal to signal incoming CLI - it then send the CLI, *then* sends the ring signals, so answering on polarity reversal would be wrong. Answer supervision on reversal polarity applies only to outgoing calls, not incoming ones. They also do a random polarity reversal most nights too - some sort of automated line testing. Eg. from my home box: Oct 7 01:40:28 NOTICE[15581] chan_zap.c: Got event 17 (Polarity Reversal)... Oct 9 03:53:06 NOTICE[19904] chan_zap.c: Got event 17 (Polarity Reversal)... Note the times... Are they just warning alarms or they starts phantom calls? As far as I know, it's BT testing the line - what they actually do, I've no idea. Fortuantely asterisk doesn't think it's an incoming call and ring all the phones.. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching *, + and # in the dialplan
On Sat, 2008-10-11 at 10:09 +0200, Benny Amorsen wrote: Tilghman Lesher [EMAIL PROTECTED] writes: exten = [0-9*#+].,... If that does not work, that is a bug and needs to be reported as such. Sadly that matches *james and 9foo... It would be nice if you could use normal regexes (e.g. with the pcre library) in extensions.conf. Full RE's are not really a good choice for extension pattern matching in the asterisk environment. People have voiced this before; but the cut-down version of RE's that the matching algorithms allow are fairly fast, both in the new and the old pattern matching algorithms. Using pcre would not be good in the dialplan. It would cut the speed of matching an extension to a fraction of what is now. It would be incompatible with the current dialplan algorithms, so you'd have to use some sort of alternate syntax. I've written at length why we use the algorithm we use now, and why pcre type implementations might seem nice, but they preclude using the current matching algorithms. Look around in the dev list archives over the last 2 years from me... Other than the above, we could invent a slightly different syntax for pcre type expressions; and you'd have to invent some sort of disambiguation for when multiple extensions might be matched, to choose the 'best' one. You'd have to forever stick with the 'check against every extension in the context for the best one' sort of algorithm, which is OK when the extension list is fairly short, but if you end up with more than 100 or so extensions in a context, you'd best go with the fast pattern matcher if you have to/want to deal with heavy call loads. I've previously offered to expand the current pattern matchers to include some useful notation available in RE's. Some constructs could be done in the current regime, but depending on which ones, will complicate the current algorithms, but are possible. Some RE features are simply incompatible with the current algorithms. murf /Benny -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about echo cancelation
On Sunday 12 October 2008 04:15:02 Olivier wrote: 2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED] Handsets use a 4-wire connection. Handsets with the the volume turned up could cause a form of echo as the microphone picks up the ear piece audio (I call this acoustic echo). Everything I said applies to 2-wire caused echo. Other types of echo is fairly uncommon and cannot be solved by normal echo canceling systems. Most echo canceling systems I've seen (mostly tellabs) only cancel echo in one direction. Which one ? Using previous example, I would say they cancel Bob's voice echo ... Said differently, echo canceller would filters out from outgoing audio signals matching signals previously heard in incoming audio (Bob's voice). I suspect all of Digium's EC systems only do echo canceling in one direction as well. Is anyone aware of EC systems working in both directions ? Pray tell, how do you echo cancel in both directions? Wouldn't that necessitate cancelling echo before it occurs on the line (sort of a white noise/pink noise kind of operation)? Seems like modelling a projectile such that when it reaches its target, the atmospheric stresses during its flight turn it into a perfect sphere (and with about the same likelihood of success). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tracking T1/PRI channel status - inbound vs. outbound
I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel x Or: Action: Status However, nothing in those results seems to reliably indicate whether the channel received an incoming call or was used to make an outgoing call. I can make educated guesses based on the value of Caller ID or Calling TON or the combination of Channel and Link, but none of these heuristics seem robust. I've even tried GetVar-ing various channel functions, to no avail. This seems like a case where a simple flag should be set somewhere, but I haven't found one. What's the most elegant way to do this? Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail issues with 1.6.0
I'm trying to get VoiceMailMain() to work properly, but it refuses. : ( I am using IMAP_STORAGE, which is functioning fine now... My voicemail.conf user line: 6000 = 1234,Brendan's Mailbox,,,[EMAIL PROTECTED]| imappassword=password 6000 = d,Brendan Martens My voicemail extension in extensions.conf: exten = 700,1,VoiceMailMain() And the output on the console: -- Executing [EMAIL PROTECTED]:1] VoiceMailMain(SIP/ 6000-489125a8, [EMAIL PROTECTED]) in new stack -- SIP/6000-489125a8 Playing 'vm-password.gsm' (language 'en') -- Incorrect password '1234' for user '6000' (context = default) -- SIP/6000-489125a8 Playing 'vm-incorrect.gsm' (language 'en') Very oddly I am having issues with updates to the files taking effect, I have changed the VoiceMailMain() to that after having previously set it to use a mailbox directly, but it still doesn't ask me what mailbox I want to check!? It seems to be stuck with mbox 6000... And before anyone asks, yes I have reloaded after making the changes, even a full restart does nothing. I am calling from ext 6000, but even when setting it to use [EMAIL PROTECTED] it still tries to check my 6000 mailbox. Regardless of some oddities in changes taking effect, the password/ user combination above is clearly correct I am baffled as to why it won't take the password, and why asterisk isn't finding my changes to the conf files. Brendan Martens ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
I have done this. Why BRIs exist in the US is beyond me. If you can, don't go with BRI. Who is the carrier. There is someone on the list that will tell you it is impossible unless you use his code, which is not true. Thanks, Steve Totaro On Mon, Oct 13, 2008 at 4:57 PM, Michael Graves [EMAIL PROTECTED] wrote: I had considered something like this as well, but was convinced to go another direction. I wrote something up about it at the time. http://www.smallnetbuilder.com/content/view/30444/84/ Michael --Original Message Text--- *From:* Wilton Helm *Date:* Mon, 13 Oct 2008 14:44:26 -0600 Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price competitively with equivalent POTS services and it makes more sense, both in terms of voice quality (4 wire digital to the PABX) and flexibility. Ideally it would allow any combination of two calls, identified by SPID. If anyone has done anything similar, or has any experience with BRI ISDN, I would appreciate input and direction. If anyone knows where documentation exists on configuring ISDN, that information would also be greatly appreciated. Asterisk has a bit of a learning curve, and ISDN BRI isn't the most widely used or covered aspect of it. BTW, I have a strong telecom background, so the theory part of it will not be a problem, only the necessary documentation to apply it to Asterisk. Thanks, Wilton Helm Embedded System Resources -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tracking T1/PRI channel status - inbound vs. outbound
On Mon, Oct 13, 2008 at 5:22 PM, Kevin DeGraaf [EMAIL PROTECTED] wrote: I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel x Or: Action: Status However, nothing in those results seems to reliably indicate whether the channel received an incoming call or was used to make an outgoing call. I can make educated guesses based on the value of Caller ID or Calling TON or the combination of Channel and Link, but none of these heuristics seem robust. I've even tried GetVar-ing various channel functions, to no avail. This seems like a case where a simple flag should be set somewhere, but I haven't found one. What's the most elegant way to do this? Thanks. -- Kevin DeGraaf AsterCRM looks like it might fit the bill. It gets all data from AMI and runs as a daemon that populates a DB. It's beta, site claims 5 calls for free but I am not sure for beta software, I would not pay for Beta software, never used it but it is new. .09Beta July 26th 2008 http://sourceforge.net/project/showfiles.php?group_id=202441 http://astercrm.blog.com/ Interesting anyways, seems the concept/code would be easy to reproduce if pricing is an issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
On Mon, 2008-10-13 at 17:37 -0400, Steve Totaro wrote: I have done this. Why BRIs exist in the US is beyond me. Much of the idea's behind ISDN are hopelesly outdated, except for one: With POTS, the analogue/Digital conversion is done some miles away, in the your local number exchange, and the distance between your phone and the exchange dictate the quality you can get. With ISDN, the conversion is done in your phone, no further deteriation. Some telco's offer a single, high bandwith channel, others dropped the second B-channel (thus offering at reduced price 1B+D) You wrote: If you can, don't go with BRI. I would agree, IF you can get a PRI-line for the same price. Some time ago, i heard that BT offered PRI-lines with a reduced number of B-channels. That would be a better alternative for BRI. hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
I had converations with both Pika and Xorcom wherein the thought that it should be possible using their interface hardware. There might be some software changes to be made in their drivers, but BRI should be usable in the US. I abandoned the idea for being more expensive when all costs are considered. Michael --Original Message Text--- From: Steve Totaro Date: Mon, 13 Oct 2008 17:37:37 -0400 I have done this. Why BRIs exist in the US is beyond me. If you can, don't go with BRI. Who is the carrier. There is someone on the list that will tell you it is impossible unless you use his code, which is not true. Thanks, Steve Totaro On Mon, Oct 13, 2008 at 4:57 PM, Michael Graves [EMAIL PROTECTED] wrote: I had considered something like this as well, but was convinced to go another direction. I wrote something up about it at the time. http://www.smallnetbuilder.com/content/view/30444/84/ Michael --Original Message Text--- From: Wilton Helm Date: Mon, 13 Oct 2008 14:44:26 -0600 Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price competitively with equivalent POTS services and it makes more sense, both in terms of voice quality (4 wire digital to the PABX) and flexibility. Ideally it would allow any combination of two calls, identified by SPID. If anyone has done anything similar, or has any experience with BRI ISDN, I would appreciate input and direction. If anyone knows where documentation exists on configuring ISDN, that information would also be greatly appreciated. Asterisk has a bit of a learning curve, and ISDN BRI isn't the most widely used or covered aspect of it. BTW, I have a strong telecom background, so the theory part of it will not be a problem, only the necessary documentation to apply it to Asterisk. Thanks, Wilton Helm Embedded System Resources -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tracking T1/PRI channel status - inbound vs. outbound
On Mon, Oct 13, 2008 at 5:53 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Mon, Oct 13, 2008 at 5:22 PM, Kevin DeGraaf [EMAIL PROTECTED] wrote: I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel x Or: Action: Status However, nothing in those results seems to reliably indicate whether the channel received an incoming call or was used to make an outgoing call. I can make educated guesses based on the value of Caller ID or Calling TON or the combination of Channel and Link, but none of these heuristics seem robust. I've even tried GetVar-ing various channel functions, to no avail. This seems like a case where a simple flag should be set somewhere, but I haven't found one. What's the most elegant way to do this? Thanks. -- Kevin DeGraaf AsterCRM looks like it might fit the bill. It gets all data from AMI and runs as a daemon that populates a DB. It's beta, site claims 5 calls for free but I am not sure for beta software, I would not pay for Beta software, never used it but it is new. .09Beta July 26th 2008 http://sourceforge.net/project/showfiles.php?group_id=202441 http://astercrm.blog.com/ Interesting anyways, seems the concept/code would be easy to reproduce if pricing is an issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Correction, daemon appears to be called astercc. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk help please
Hi, I am new user on asterisk (for that matter linux) and i have lot of embedded programming experience. We have a new project from our client, to design a box that takes the telelphone line as input and route the line to the respective user with different ring tones. The box should be programmed by the users with buttons. Features. 1. I should be able to store some .wav files for different ringtones and for voicemail 2. Should be programmable by user. Questions: 1. Need to know the cpu that can support asterisk for this type of application 2 What else do we need on the box to support the application 3 Any pointers related to this would be really appreciated. Thanks Ram On Mon, Oct 13, 2008 at 4:55 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Mon, Oct 13, 2008 at 5:53 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Mon, Oct 13, 2008 at 5:22 PM, Kevin DeGraaf [EMAIL PROTECTED]wrote: I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel x Or: Action: Status However, nothing in those results seems to reliably indicate whether the channel received an incoming call or was used to make an outgoing call. I can make educated guesses based on the value of Caller ID or Calling TON or the combination of Channel and Link, but none of these heuristics seem robust. I've even tried GetVar-ing various channel functions, to no avail. This seems like a case where a simple flag should be set somewhere, but I haven't found one. What's the most elegant way to do this? Thanks. -- Kevin DeGraaf AsterCRM looks like it might fit the bill. It gets all data from AMI and runs as a daemon that populates a DB. It's beta, site claims 5 calls for free but I am not sure for beta software, I would not pay for Beta software, never used it but it is new. .09Beta July 26th 2008 http://sourceforge.net/project/showfiles.php?group_id=202441 http://astercrm.blog.com/ Interesting anyways, seems the concept/code would be easy to reproduce if pricing is an issue. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Correction, daemon appears to be called astercc. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
The documentation is in my head, two solid days worth. The issue is the SPID code that Marcin Pyco claimed he had the only code, and way to make it work in the US.. You may need this code if you are using SPIDs to route calls. In my situation, they were just a hunt group, two BRIs, and I was tasked with adding a quad port Sangoma analog card. Absolutely NO difference in audio, but talk about a mish mash of equipment. Luckily Sangoma drivers for Zaptel 1.4 do not require Zaptel to be patched. It absolutely refused to work with 1.2 so it became my first 1.4 installation out of necessity, I am sure 1.2 didn't work because of conflicting patches (BRIStuff and Sangoma) That is why Xorcom was so happy to help me with a US BRI, and I just thought Tzafrir was a nice guy trying to help out... Marcin Pyco claimed that BRI would not work without his code in the US and went so far as to call me a liar. I proved him wrong, but he is not very good at admitting he is wrong, he blamed Verizon rather than apologizing. He is very good at calling people liars but not so good at apologizing and admitting he is wrong. Whatever the rub, using BRIStuff, Zaptel 1.4.X and a Junghanns' card or knock-off (and even Sangoma drivers), it will work with Verizon. I have pages upon pages of all the emails and IRC chats where I am called a liar, and where Tzafrir admits his true motives (to his credit). And finally the revelation that you do not need any additional code for SPIDs (at least with Verizon) in the US, and around here everyone resells Verizon anyways. One thing to note is that inbound calls work immediately when the spans come up BUT it takes ten to fifteen minutes for outbound calls to work. I am not sure if the time starts at loading qozap or Asterisk but it works beyond a shadow of a doubt, so don't pay for code that makes it work. I am convinced that the conversations you had with Xorcom and probably Pika (since Marcin works or worked there (LinkedIN)) came as a direct result of my work. Anyways, in this area, everything is close to a CO and I BET that calling from a regular phone, you could never guess which is ISDN and which is POTS, unless you cheat somehow, but not by voice quality. I am not sure why OP thinks that two pair for voice is better than two unless he is afraid of echo, which was absolutely no issue with the Sangoma cards with onboard EC. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Mon, Oct 13, 2008 at 5:55 PM, Michael Graves [EMAIL PROTECTED] wrote: I had converations with both Pika and Xorcom wherein the thought that it should be possible using their interface hardware. There might be some software changes to be made in their drivers, but BRI should be usable in the US. I abandoned the idea for being more expensive when all costs are considered. Michael --Original Message Text--- *From:* Steve Totaro *Date:* Mon, 13 Oct 2008 17:37:37 -0400 I have done this. Why BRIs exist in the US is beyond me. If you can, don't go with BRI. Who is the carrier. There is someone on the list that will tell you it is impossible unless you use his code, which is not true. Thanks, Steve Totaro On Mon, Oct 13, 2008 at 4:57 PM, Michael Graves [EMAIL PROTECTED] wrote: I had considered something like this as well, but was convinced to go another direction. I wrote something up about it at the time. *http://www.smallnetbuilder.com/content/view/30444/84/* Michael --Original Message Text--- *From:* Wilton Helm *Date:* Mon, 13 Oct 2008 14:44:26 -0600 Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price competitively with equivalent POTS services and it makes more sense, both in terms of voice quality (4 wire digital to the PABX) and flexibility. Ideally it would allow any combination of two calls, identified by SPID. If anyone has done anything similar, or has any experience with BRI ISDN, I would appreciate input and direction. If anyone knows where documentation exists on configuring ISDN, that information would also be greatly appreciated. Asterisk has a bit of a learning curve, and ISDN BRI isn't the most widely used or covered aspect of it. BTW, I have a strong telecom background, so the theory part of it will not be a problem, only the necessary documentation to apply it to Asterisk. Thanks, Wilton Helm Embedded System Resources -- Michael Graves mgravesat*mstvp.com* *http://blog.mgraves.org* o713-861-4005 c713-201-1262 *sip:[EMAIL PROTECTED] [EMAIL PROTECTED]* skype mjgraves fwd 54245 ___ --
Re: [asterisk-users] ISDN
I'm in the process of setting up Asterisk in a SOHO environment using = ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID = is used for the business and the other is used for personal. The = circuit already exists, but is presently being interfaced to POTS phones = via a TA. This configuration is not very common in the US, but we are fortunate = that our LEC offers it price competitively with equivalent POTS services = and it makes more sense, both in terms of voice quality (4 wire digital = to the PABX) and flexibility. Ideally it would allow any combination of two calls, identified by SPID. If anyone has done anything similar, or has any experience with BRI = ISDN, I would appreciate input and direction. If anyone knows where documentation exists on configuring ISDN, that = information would also be greatly appreciated. Asterisk has a bit of a = learning curve, and ISDN BRI isn't the most widely used or covered = aspect of it. BTW, I have a strong telecom background, so the theory = part of it will not be a problem, only the necessary documentation to = apply it to Asterisk. The one solution I've heard, on and off again, that works with Asterisk here in the US is the Eicon Diva cards. There are other solutions. Where I am, we're unreasonably close to a local radio conglomerate that has a number of high power antennas. We found early on that RF interference was a killer, which caused me to run a lot of our telecom and data wiring in conduit. Unfortunately, we discovered that POTS lines were a hell of a mess when connected to anything more complex than a phone or two. Lots of RF interference. Church radio music on Sundays, even. So, we brought our lines in on BRI, which we've used for data and voice elsewhere. Being eternally frustrated with the lack of ISDN support after maybe 2000 here in the US (we have a bunch of interesting ISDN gear from the 90's!), I set out to see what I could do to interface BRI to Asterisk. I *didn't* go the Eicon route, because at the time it was considered relatively unreliable. Instead, we picked up an Adtran Atlas 550, which can handle ISDN BRI, PRI, POTS, etc. We have been using the Atlas as a translator to convert BRI to T1, which works moderately well, but we've seen some issues, mostly in the capabilities of the Adtran (such as an inability to select the desired SPID/DN for outgoing calls). The Adtran has some other amazing capabilities, such as providing FXO/FXS ports, and even ISDN BRI ports for other devices we'd liked hooked into our PBX. Despite that, I'd love to see an ISDN BRI solution for the US. I might be willing to test the Eicon Diva Server card... hm. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line
On Sun, Oct 12, 2008 at 11:18:40AM +0100, Gordon Henderson wrote: On Thu, 9 Oct 2008, Mike wrote: I'm guessing this lamp is on an ordinary analogue phone you have? Yeah, this is a bog standard 9 quid analogue phone. OK. A bit convoluted this as I'm not local to the PBX, but an IAX trunk, another asterisk and a SIP phone away from it - however I'm looking at verbose console output on the original asterisk box which has an OpenVox card (TDM400 clone) so: Telewest line - TDM - Asterisk - IAX trunk - Asterisk - SIP phone I call into the site on their Telewest landline from my mobile (O2). My phone rings, which is to be expected. I don't answer it, but hangup my mobile. I can't tell if telewest have dropped the line immediately, but from asterisk point of view, the line stays open for about another 2-3 seconds, then hangs up and asterisk detects it and stops the phone ringing. As far as I can tell, this is all perfectly normal, and it's what I see with BT lines too. If I answer it, that's fine too. If I then hangup the mobile, the (analogue) line hangs up almost immediately, this is detected almost immediately by asterisk and it clears down the call. So that's more or less what I'm expecting. Going the other way: Dialled out from my SIP phone to my mobile - when I hungup the mobile the call dropped almost instantly and my SIP phone hungup. So again, that's more or less what I expect. Thanks for doing that. Do you mind showing me your zapata.conf? I've heard that Telewest used whatever switch they could get their hands on at the time when they were building their network, and that different regions might well have different equipment in the TW exchanges. More complicated by them buying up the local cable co's (eg. Eurobell in the Plymouth area where this line is), then being sucked into the ntl: monster and now virgin media. Yeah, I gather all the many, small telcos doing cable all eventually merged together. I'm trying to get hold of a Telewest engineer to find out what is going on. Naturally this is proving challenging... I'm trying to work out what to expect from the line and see if that is consistent with what I am seeing. Once I know what the phone line is meant to do, then I can work out if it is doing and what I can do with Asterisk to accomodate it. Good luck! Gordon Mike. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching *, + and # in the dialplan
Steve Murphy [EMAIL PROTECTED] wrote: People have voiced this before; but the cut-down version of RE's that the matching algorithms allow are fairly fast, both in the new and the old pattern matching algorithms. Steve Your explanation is clear and it seems like a good design choice to exclude support for regular expressions, but what seems odd (maybe a bug in fact) is the specific exclusion of characters +, # and *. It sounds like you're saying: exten = [0-9*#+].,... is invalid, therefore not a bug, and that only numeric parameters such as: exten = [0-6].,... would be valid. If this is correct could you please explain the proper way to match any extension beginning with + such as '+13129842314' without also matching: 'i' Thanks for your input Steve! -Karl What extension the following: '3129842314' '*989' '+13129842314' BUT does not match: 'i' 'james' I'd like to see a wildchard character that matches Can support for those characters be added without ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
On Mon, Oct 13, 2008 at 04:55:01PM -0500, Michael Graves wrote: I had converations with both Pika and Xorcom wherein the thought that it should be possible using their interface hardware. There might be some software changes to be made in their drivers, but BRI should be usable in the US. Or actually: I suppose that now that Asterisk finally knows that spans can be of size 2B+1D (as of 1.4.22), chan_dahdi will support US BRI will with any BRI device that has a Zaptel/DAHDI driver. That should be either ours (Xorcom) BRI module of the Astribank, Junghanns quad/octo/duo BRI cards and compatible, or the simple HFC-S -based single port PCI cards. Sangoma A500 cards should have Zaptel drivers as well. However this is based on quite a few cases and mostly remains to be tested. (The relevant patch to Asterisk is trivial to apply to earlier versions of 1.4 / 1.2) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching *, + and # in the dialplan
exten = +13129842314,1,Noop(Happy match!) or exten = _+1NXXNXX,1,Noop(Happier match!) Karl Fife wrote: Steve Murphy [EMAIL PROTECTED] wrote: People have voiced this before; but the cut-down version of RE's that the matching algorithms allow are fairly fast, both in the new and the old pattern matching algorithms. Steve Your explanation is clear and it seems like a good design choice to exclude support for regular expressions, but what seems odd (maybe a bug in fact) is the specific exclusion of characters +, # and *. It sounds like you're saying: exten = [0-9*#+].,... is invalid, therefore not a bug, and that only numeric parameters such as: exten = [0-6].,... would be valid. If this is correct could you please explain the proper way to match any extension beginning with + such as '+13129842314' without also matching: -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cli commands missing
On Mon, Oct 13, 2008 at 06:51:47PM +0200, Karsten Wemheuer wrote: Hi Eric, Am Sonntag, den 12.10.2008, 18:06 -0700 schrieb Eric Fort: resolve.conf and dns is working. The problem persists. /var/log/asterisk/messages shows a few notices and warnings on res_smdi.c, res_musiconhold.c, and usbradio.c. when I disable loading of these in modules.conf asterisk crashes on load. Eric try to start asterisk in foreground. First stop it and then start it on the shell: asterisk -vv Actually often many v-s flood you with messages and hide the errors. asterisk -U asterisk -c But then again, this information should already be in the logs. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4
Yes, I certainly applied the patch in http://ftp.iq-labs.net/queue_log-1.4/asterisk_queue_log_realtime_1.4.19. patch Just to double-check, there is only one patch in this URL which is main/logger.c By the way, did you see anything wrong with my config files? /etc/asterisk/extconfig.conf [settings] queue_log = mysql,db1 /etc/asterisk/res_mysql.conf [general] dbhost = localhost dbname = db1 dbuser = user dbpass = password dbport = 3306 dbsock = /var/lib/mysql/mysql.sock -Original Message- From: Atis Lezdins [mailto:[EMAIL PROTECTED] Sent: Monday, 13 October 2008 8:02 PM To: Lee, John (Sydney) Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4 Hi John, On Mon, Oct 13, 2008 at 9:51 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: http://ftp.iq-labs.net/queue_log- 1.4/asterisk_queue_log_realtime_1.4.19.patch Haven't you forgotten this one? ;) if you have applied everything correctly - queue_log file shoudln't have any more lines (except init when restarting asterisk). Regards, Atis This uses standardized realtime/mysql library from asterisk addons. For it to support SQL inserts in 1.4, you would also need to apply both patches from (1 for asterisk, another for asterisk-addons) http://ftp.iq-labs.net/realtime_store_destroy-1.4/ This will later allow you to upgrade to 1.6 and having everything working without patching. I have patched in asterisk 1.4 . main/logger.c . include/asterisk/config.h . main/config.c I have patched in asterisk-addons 1.4 . res/res_config_mysql.c I have re-installed asterisk and asterisk-addons. I created a database called db1 and in there created a table called queue_log as per instruction http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL I changed /etc/asterisk/extconfig.conf to add the following line: [settings] queue_log = mysql,db1 I changed /etc/asterisk/res_mysql.conf to add the following: [general] dbhost = localhost dbname = db1 dbuser = user dbpass = password dbport = 3306 dbsock = /var/lib/mysql/mysql.sock 1) However, whenever I perform an agent login, no row is written to table queue_log. I checked /var/log/asterisk/queue_log and a new entry is written there. 2) I set debug to 10 on the console in asterisk and re-did the test but there were no error messages in /var/log/asterisk/messages. 3) I set debug on in mysqld and there are no information for inserting into table queue_log, except the cdr logging as below. Tcp port: 0 Unix socket: (null) Time Id CommandArgument 081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1 2 Connect [EMAIL PROTECTED] on db1 081013 16:00:32 1 Query INSERT INTO cdr_log ... 081013 16:01:42 1 Query INSERT INTO cdr_log ... Is there anyone who can help me? -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4
if you have applied everything correctly - queue_log file shoudln't
have any more lines (except init when restarting asterisk).
Thanks Atis.
I see what you are saying. In the patch for logger.c,
The code to write to mysql is there except that we need to perform
ast_check_realtime(queue_log).
I guess ast_check_realtime() is looking into extconfig.conf and
searching for
queue_log = mysql,db1
which is there in my extconfig.conf already.
Can any Asterisk developers enlighten me on this?
void ast_queue_log(const char ...)
{
+ char qlog_msg[8192];
+ char time_str[16];
+
+ if (ast_check_realtime(queue_log)) {
va_start(ap, fmt);
+ vsnprintf(qlog_msg, sizeof(qlog_msg), fmt, ap);
va_end(ap);
+
+ snprintf(time_str, sizeof(time_str), %ld,
(long)time(NULL));
+ ast_store_realtime(queue_log, time, time_str,
+ callid, callid,
+ queuename, queuename,
+ agent, agent,
+ event, event,
+ data, qlog_msg,
+ NULL);
+ } else {
+ if (qlog) {
+ AST_LIST_LOCK(logchannels);
+ va_start(ap, fmt);
+ fprintf(qlog, %ld|%s|%s|%s|%s|,
(long)time(NULL), callid, queuename, agent, event);
[...]
+ }
}
-Original Message-
From: Atis Lezdins [mailto:[EMAIL PROTECTED]
Sent: Monday, 13 October 2008 8:02 PM
To: Lee, John (Sydney)
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] realtime queue_log to mySQL backport to
1.4
Hi John,
On Mon, Oct 13, 2008 at 9:51 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
http://ftp.iq-labs.net/queue_log-
1.4/asterisk_queue_log_realtime_1.4.19.patch
Haven't you forgotten this one? ;)
if you have applied everything correctly - queue_log file shoudln't
have any more lines (except init when restarting asterisk).
Regards,
Atis
This uses standardized realtime/mysql library from asterisk addons.
For it to support SQL inserts in 1.4, you would also need to apply
both patches from (1 for asterisk, another for asterisk-addons)
http://ftp.iq-labs.net/realtime_store_destroy-1.4/
This will later allow you to upgrade to 1.6 and having everything
working without patching.
I have patched in asterisk 1.4
. main/logger.c
. include/asterisk/config.h
. main/config.c
I have patched in asterisk-addons 1.4
. res/res_config_mysql.c
I have re-installed asterisk and asterisk-addons.
I created a database called db1 and in there created a table called
queue_log as per instruction
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
I changed /etc/asterisk/extconfig.conf to add the following line:
[settings]
queue_log = mysql,db1
I changed /etc/asterisk/res_mysql.conf to add the following:
[general]
dbhost = localhost
dbname = db1
dbuser = user
dbpass = password
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
1) However, whenever I perform an agent login, no row is written to
table queue_log. I checked /var/log/asterisk/queue_log and a new
entry
is written there.
2) I set debug to 10 on the console in asterisk and re-did the test
but
there were no error messages in /var/log/asterisk/messages.
3) I set debug on in mysqld and there are no information for
inserting
into table queue_log, except the cdr logging as below.
Tcp port: 0 Unix socket: (null)
Time Id CommandArgument
081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1
2 Connect [EMAIL PROTECTED] on db1
081013 16:00:32 1 Query INSERT INTO cdr_log ...
081013 16:01:42 1 Query INSERT INTO cdr_log ...
Is there anyone who can help me?
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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[asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. * I need to get a formatted calls report for the administrators to charge the users. I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. Thanks a lot. -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
With ISDN, the conversion is done in your phone Exactly. Or in the case of Asterisk, it is a 4 wire digital right into the switch--no degradation. Even converting back and forth between analog and digital multiple times compromises quality. Try doing a dial-up modem across such a path. The best you will get is 20 - 30 K. IF you can get a PRI-line for the same price. Not to mention that the interfaces for PRI are about five times as expensive. I'm not sure why. It doesn't seem like it ought to take a lot of electronics to break down the bit stream. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
I have done this. Good Why BRIs exist in the US is beyond me. I'm not sure why you say that. It is the only way I know of two get two digital voice grade circuits at prices competitive with POTS. The better question is why the LECs used such poor judgment when they introduced this. Most were charging outrageous prices that had no technical justification and per minute usage fees. They destroyed the market before it got off the ground. I don't use it for data any more now that I have DSL, but it is still a very viable voice channel. If you can, don't go with BRI. Why? (Not that I don't already have it and have been using it for seven years.) Who is the carrier. Qwest (formerly US West or US Worst as some used to call it). So do you have some information about what you did or where to get configuration information? Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN
The card I have has no name but is based on the Winbond W6692CF chip and ships with RVS, which I think is for Windows and of no use to me. I'm not sure about whether it is supported by DAHDI or not. Wilton___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
Rodolfo Alcazar Portillo wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. Asterisk is more featured than Panasonic, but you must to know Asterisk to convince your executives ;-) What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... Probably you are sending dtmf signals inband. Try outband. For the echo, try to change the FXO/FXS impedance, and/or playing with the rx and tx gains. I assume that do you have echo cancelling enable in both SPA. * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. See account codes. You can start here: http://www.voip-info.org/wiki-Asterisk+Billing * I need to get a formatted calls report for the administrators to charge the users. See same link, or google for billing I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. The link to start: http://www.voip-info.org Thanks a lot. De nada Jorge ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging and Asterisk
Can somebody please give a pointer to a complete neophyte (like me) on text messaging, what product can I use to send and automatic text message to a cell phone from within the asterisk dialplan? (the part of the dialplan I have down, the part of the text message no) Thanks C. Savinovich 1) Email-to-SMS gateways are plentiful, and most carriers have them. Google will give more details, but WikiPedia has good data to start: http://en.wikipedia.org/wiki/SMS_gateway 2) Connect a GSM phone to your *NIX system. There are several software packages out there to do this via serial port. Command-line tools then can send SMS messages via the local phone. Put a System() call in your dialplan, or (better) use an AGI. 3) Use a service provider that does SMS gatewaying. Much more reliable than the first two methods. One example firm I've used with very easy startup and several APIs including email: Clickatell.com. Others: http://www.160characters.org/pages.php?action=viewpid=7 JT -- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials without making a call?
At 9:37 AM -0700 2008/10/13, Eric Chamberlain wrote: On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote: Eric Chamberlain wrote: Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making outbound calls. In the case of a server like Asterisk, wouldn't sending a register disrupt the flow of inbound calls until the UA that normally handles inbound calls re-registers? Are you using the same credentials as existing extensions to make calls from different extensions? That would seem to be a particularly bad idea. You should be configuring /one/ sip extension per SIP phone. Those extensions that handle outgoing calls only could be put in a different number range, or have a letter prefixed or suffixed to the extension, but you should /not/ be using one configured extension for two different purposes. We're developing the client and don't have control over the server, which may or may not be Asterisk. Adding extra extensions isn't possible. Can OPTION packets be used to verify authentication? I've never seen a system that I can recall that uses authentication on OPTIONS, since OPTIONS aren't generally considered to be secure. I'm sure some exist, but it'd be a poor test beacon. I can't think of anything off the top of my head that wouldn't create some sort of record. How about trying to call your own AOR? At least it under most circumstances would be no cost, and you'd be able to ignore your own call or even better, you'd get a LOOP DETECTED message which would only come after you've authenticated. JT -- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Budge Tones pick up wrong calls
Paul Douglas Franklin wrote: When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For example, 2501 might respond to the calls for 2518. After a reboot, it might decide to respond to 2501 as it should. Or it might respond to 2536. The phone it responds for will not respond. I have seen this with Polycom phones. In my case the problem turned out to be because there were several phones behind NAT and the NAT router got a little confused. The only solution I could find was to have the phones use different ports - ie. 5060, 5061, 5062. When they all shared 5060 the NAT router was unable to keep track of where an incoming call should be routed to. Hope this helps, Trevor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra phones and dns srv records
Tom Moore wrote: Hi guys, Does the Aastra line of phones work with dns srv records? I'm trying to get my 8133i to do this and in the settings it asks for ip addresses of registration and proxy servers. Does this mean that it will not just let me put the domain name in like other devices I have and then do fail over to other servers when needed? If these phones do not what phones do? Have you tried putting in a domain name to see what happens? I use domain names on all my Aastra phones. I don't think they have SRV support, bu they certainly do have DNS support. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
