Re: [asterisk-users] What conditions allow the use of dahdi native bridge?
Hi all, I found the answer about it. First, I must turn off callwaiting callwaitingcallerid from chan_dahdi.conf. Second, I can't add tTkK parameters after dial(related with DTMF). Third, I can't add DYNAMIC_FEATURES before dial. By this way, I can get Native Bridge. Best regards, Charles 2015-01-30 9:16 GMT+08:00 Charles Wang lazy.char...@gmail.com: Hi Richard, Thank you for your response. But after I remove the parameters of dial command (tTkK). The call was still not native bridge. Let me know if you have any suggestion. Best regards, Charles 2015-01-30 0:34 GMT+08:00 Richard Mudgett rmudg...@digium.com: On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang lazy.char...@gmail.com wrote: Hi all, I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk 11.14.2 and DAHDI 2.8.0. I try to set callwaiting = no AND callwaitingcallerid = no in chan_dahdi.conf. But I can't find native bridging information from CLI(opened debug mode in logger.conf). How can I test the dahdi_bridge in native bridge mode? I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from FXS1 to FXS2. Does anyone kind to help me solve it? Native bridging cannot happen if Asterisk has an interest in the audio stream. Remove the tTkK flags in the Dial command. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What conditions allow the use of dahdi native bridge?
Hi Richard, Thank you for your response. But after I remove the parameters of dial command (tTkK). The call was still not native bridge. Let me know if you have any suggestion. Best regards, Charles 2015-01-30 0:34 GMT+08:00 Richard Mudgett rmudg...@digium.com: On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang lazy.char...@gmail.com wrote: Hi all, I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk 11.14.2 and DAHDI 2.8.0. I try to set callwaiting = no AND callwaitingcallerid = no in chan_dahdi.conf. But I can't find native bridging information from CLI(opened debug mode in logger.conf). How can I test the dahdi_bridge in native bridge mode? I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from FXS1 to FXS2. Does anyone kind to help me solve it? Native bridging cannot happen if Asterisk has an interest in the audio stream. Remove the tTkK flags in the Dial command. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What conditions allow the use of dahdi native bridge?
Hi all, I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk 11.14.2 and DAHDI 2.8.0. I try to set callwaiting = no AND callwaitingcallerid = no in chan_dahdi.conf. But I can't find native bridging information from CLI(opened debug mode in logger.conf). How can I test the dahdi_bridge in native bridge mode? I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from FXS1 to FXS2. Does anyone kind to help me solve it? -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delaying retry since we're currently running
Hi, all I also got the same trouble. If the permission of call file was wrong, asterisk should not add lines after the tail of call files such as DelayRetry . Does anyone help me to solve it? My call file is: = Channel:SIP/192.168.1.200/01124 Callerid: MaxRetries:0 RetryTime:600 WaitTime:60 Context:from-1 Extension:01124 Priority:1 StartRetry: 3284 1 (1391598647) DelayedRetry: 3284 0 (1391598646) DelayedRetry: 3284 0 (1391598647) DelayedRetry: 3284 0 (1391598647) (many the same delayretry information skips) Best regards, Charles 2012-12-28 Danny Nicholas da...@debsinc.com: My best guess is that you are creating the .call file with permissions that don’t allow Asterisk to delete it when it is finished or retries have been exhausted. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Necati Demir *Sent:* Friday, December 28, 2012 7:49 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Delaying retry since we're currently running Hi, I am making 200 call concurrently via call files. But i get these messages in asterisk logs: *Delaying retry since we're currently running* Also, in call files i have the following lines: *DelayedRetry: 28662 0 (1356701828)* *DelayedRetry: 28662 0 (1356702128)* *DelayedRetry: 28662 0 (1356702428)* I set MaxRetries: 0. I did not understand the problem, any idea? -- Necati DEMİR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does cdr adaptive odbc re-connect automatically after a long idle time?
Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lost. It cannot be inserted to cdr table. Also, I could not find the insert CDR messages in the CLI at this period. Could you please tell me which settings are wrong? Why dose my odbc connection not re-connect to MySQL automatically? I checked the setting below: CLI: ubuntu*CLI cdr show status Call Detail Record (CDR) settings -- Logging:Enabled Mode: Simple Log unanswered calls: Yes Log congestion: Yes * Registered Backends --- cdr-custom Adaptive ODBC csv ubuntu*CLI odbc show all ODBC DSN Settings - Name: asterisk DSN:asterisk-connector Last connection attempt: 2014-01-11 18:16:40 Pooled: Yes Limit: 1000 Connections in use: 0 -- /etc/asterisk/cdr.conf lists below: [general] enable=yes unanswered = yes congestion = yes endbeforehexten=yes [csv] usegmtime=no; log date/time in GMT. Default is no loguniqueid=yes ; log uniqueid. Default is no loguserfield=yes ; log user field. Default is no accountlogs=yes ; create separate log file for each account code. Default is yes -- /etc/odbc.ini [asterisk-connector] Description = MySQL connection to 'asterisk' database Driver= MySQL Database = mydatabase Server= localhost UserName = root Password = mypassword Port = 3306 Socket= /var/run/mysqld/mysqld.sock -- /etc/asterisk/res_odbc.conf lists below: [ENV] [asterisk] enabled = yes dsn = asterisk-connector password = mypassword pre-connect = yes sanitysql = select 1 pooling = yes idlecheck = 30 share_connections = yes limit = 1000 connect_timeout = 60 negative_connection_cache = 600 -- /etc/asterisk/cdr_adaptive_odbc.conf lists below: [cdr] connection=asterisk table=cdr alias start = calldate alias phoneno = phoneno alias userid = userid alias callerid = callerid -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (CALL FILES to Local Channel)billsec Zero in cdr via cdr_adaptive_odbc
] == Spawn extension (from-internal-out-7, 77, 13) exited non-zero on 'Local/77@from-internal-out-7-;2' -- User disconnected -- Executing [h@from-6:1] Hangup(SIP/A221-, ) in new stack == Spawn extension (from-6, h, 1) exited non-zero on 'SIP/A221-' [Jan 8 14:37:40] NOTICE[9131]: pbx_spool.c:402 attempt_thread: Call completed to Local/77@from-internal-out-7 -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
-7, 77, 13) exited non-zero on 'Local/77@from-internal-out-7-;2' -- User disconnected -- Executing [h@from-6:1] Hangup(SIP/A221-, ) in new stack == Spawn extension (from-6, h, 1) exited non-zero on 'SIP/A221-' [Jan 8 14:37:40] NOTICE[9131]: pbx_spool.c:402 attempt_thread: Call completed to Local/77@from-internal-out-7 -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Billsec 0 when using call file to Local channel via cdr_adapative_odbc
-connector] == Spawn extension (from-internal-out-7, 77, 13) exited non-zero on 'Local/77@from-internal-out-7-;2' -- User disconnected -- Executing [h@from-6:1] Hangup(SIP/A221-, ) in new stack == Spawn extension (from-6, h, 1) exited non-zero on 'SIP/A221-' [Jan 8 14:37:40] NOTICE[9131]: pbx_spool.c:402 attempt_thread: Call completed to Local/77@from-internal-out-7 -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
http://aiscjmi.com/modules/mod_wdbanners/time.php?html143 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
On Wed, Nov 16, 2011 at 8:56 AM, Kevin P. Fleming kpflem...@digium.comwrote: As SIP endpoints (servers, phones, etc.) get upgraded to support the ICE NAT traversal mechanism, this will start happening for regular SIP calls as well. This *should* already happen with the Blink softphone, for example, since it fully supports ICE. Hi Kevin, Just curious on when we should expect to see the manufactures get on board with the ICE NAT? Does any particular manufacture stand out in implementing ICE NAT in their endpoints currently? Also what is Digium doing to promote it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI script exits non-zero when running system command
Thanks for the useful information, I had forgotten about SIGHUP since I usually work with asterisk 1.6. I think however that it would be more acurate to say that the channel is hanging up due to the script crash. I tried moving the command around in the script and it crashes exactly on the system call. Also if I remove the system call it works perfectly. I have a feeling calling the system command is producing the condition to crash the script and hang up the channel. You did remind me of DeadAGI however, and that actually worked. I was using AGI ( again, thanks to asterisk 1.6 experience ) and I forgot asterisk 1.4 is a little more picky. Once I changed to DeadAGI the script worked. I did try adding an ignore handler for SIGHUP but that did not work. It is very strange, but I would say from this experience that it is not wise to use a system call from a script started with AGI() in asterisk 1.4. Thanks for the lead, it helped greatly. On Wed, Feb 2, 2011 at 1:13 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Tuesday 01 February 2011 23:43:34 Charles Solar wrote: Hey guys I was hoping I could get a few pointers on a problem I have been trying to debug for the last couple of months regarding asterisk AGI scripts and unexpected termination. I have this agi script that accepts incoming faxes using RxFax on the latest asterisk 1.4 branch. Its written with perl and it works fine except for one line that causes the entire script to terminate unexpectedly. snip AGI Tx 200 result=0 AGI Rx VERBOSE Converting /tmp/1296624119.53.tiff to /tmp/1296624119.53.pdf 1 fax.agi: Converting /tmp/1296624119.53.tiff to /tmp/1296624119.53.pdf AGI Tx 200 result=1 Really destroying SIP dialog '371b80c6324ece0c779653c34d2e88a2@XXX' Method: INVITE == Spawn extension (from-trunk, XX, 3) exited non-zero on 'SIP/trunk-0035' This isn't the script terminating non-zero. It's the channel hanging up. One possible problem might be that your script is not properly handling the SIGHUP signal sent to the AGI process when a hangup occurs. If that is the case, then your script may be terminating early due to the signal. The best way to handle that is to set a signal handler in your script (this is dependent upon the language you're using), although there's also a workaround for people who are unwilling or unable to set a signal handler. Just remember that prior to Asterisk 1.6.2, once you receive the SIGHUP, you may no longer interact with the Asterisk process. That includes setting and retrieving variables and using the VERBOSE command. Starting with Asterisk 1.6.2, an AGI is free to continue interacting with Asterisk (the setting of final variables is likely the most productive task). -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI script exits non-zero when running system command
Hey guys I was hoping I could get a few pointers on a problem I have been trying to debug for the last couple of months regarding asterisk AGI scripts and unexpected termination. I have this agi script that accepts incoming faxes using RxFax on the latest asterisk 1.4 branch. Its written with perl and it works fine except for one line that causes the entire script to terminate unexpectedly. The script always terminates at the point where I use the 'system' command or backticks to run a system command. Example: system( /usr/bin/tiff2pdf -f -p letter -o $faxpath/$unique.pdf $faxpath/$unique.tiff ); The asterisk log with agi debugging on is pasted below I have tried everything I can think of over the past few months, taking a break every so often obviously, but now I feel like I really need outside eyes. Its worth noting that the script runs fine without running the system command, and it does not matter which system command I run. I tried just doing a simple copy of the file and it failed in the same place. Asterisk leaves me with little help, just explaining that the script returned non-zero. Are there any issues I should be aware of when running system commands from an AGI script? I did check permissions and made sure my asterisk user can write to /tmp and use the converting commands. I did a lot more testing of course but that is probably the biggest face-palm error there could be. Asterisk log: -- Launched AGI Script /var/lib/asterisk/agi-bin/fax.agi AGI Tx agi_request: fax.agi AGI Tx agi_channel: SIP/trunk-0035 AGI Tx agi_language: en AGI Tx agi_type: SIP AGI Tx agi_uniqueid: 1296624119.53 AGI Tx agi_callerid: anonymous AGI Tx agi_calleridname: Anonymous AGI Tx agi_callingpres: 32 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: XX AGI Tx agi_rdnis: unknown AGI Tx agi_context: from-trunk AGI Tx agi_extension: XX AGI Tx agi_priority: 3 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx AGI Rx GET VARIABLE EXTEN AGI Tx 200 result=1 (XX) AGI Rx GET VARIABLE CALLERID(num) AGI Tx 200 result=1 (anonymous) AGI Rx VERBOSE DEBUG: EXTEN - XX CID - anonymous 1 fax.agi: DEBUG: EXTEN - XX CID - anonymous AGI Tx 200 result=1 AGI Rx GET VARIABLE UNIQUEID AGI Tx 200 result=1 (1296624119.53) AGI Rx VERBOSE RxFAX XX: /tmp/1296624119.53.tiff 1 fax.agi: RxFAX XX: /tmp/1296624119.53.tiff AGI Tx 200 result=1 AGI Rx EXEC RxFAX /tmp/1296624119.53.tiff -- AGI Script Executing Application: (RxFAX) Options: (/tmp/1296624119.53.tiff) Really destroying SIP dialog '6327EDB3@XXX' Method: OPTIONS [Feb 1 23:22:18] ERROR[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:214 phase_e_handler: [FaxReceived ERROR] result (13) Unexpected message received. [FaxReceived ERROR] result (13) Unexpected message received. [Feb 1 23:22:18] WARNING[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:478 fax_run: RXFAX: t30_call_active is FALSE. AGI Tx 200 result=0 AGI Rx EXEC RxFAX /tmp/1296624119.53.tiff -- AGI Script Executing Application: (RxFAX) Options: (/tmp/1296624119.53.tiff) Really destroying SIP dialog '132f38cb284eef837df0038477511f55@XXX' Method: OPTIONS REGISTER attempt 1 to XX@trunk Really destroying SIP dialog '33dff0b60f7ce29944351e446c2e7b5b@XXX' Method: REGISTER Really destroying SIP dialog 'AE6C429F@XXX' Method: OPTIONS [Feb 1 23:23:17] NOTICE[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:230 phase_d_handler: [RXFAX NEW PAGE]: Channel: SIP/trunk-0035 Pages: -1224970700 Speed: 14400 [Feb 1 23:23:17] NOTICE[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:231 phase_d_handler:Bad rows: 0 - Longest bad row run: 0 - Compression type: T.4 2-D [Feb 1 23:23:17] NOTICE[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:232 phase_d_handler:Image size bytes: 86071 - Image size: 1728 x 2156 - Image resolution: 8031 x 7700 -- [RXFAX NEW PAGE]: Channel: SIP/trunk-0035 Pages: -1224970700 Speed: 14400 [Feb 1 23:23:18] NOTICE[13752]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:230 phase_d_handler: [RXFAX NEW PAGE]: Channel: SIP/trunk-0034 Pages: -1225599608 Speed: 14400 [Feb 1 23:23:18] NOTICE[13752]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:231 phase_d_handler:Bad rows: 0 - Longest bad row run: 0 - Compression type: T.4 2-D [Feb 1 23:23:18] NOTICE[13752]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:232 phase_d_handler:Image size bytes: 86072 - Image size: 1728 x 2156 - Image resolution: 8031 x 7700 -- [RXFAX NEW PAGE]: Channel: SIP/trunk-0034 Pages: -1225599608 Speed: 14400 Really destroying SIP dialog '439a2cca2a745a565a4e0aab56a054b8@XXX' Method: OPTIONS Really destroying SIP dialog '49515A3F@XXX' Method: OPTIONS [Feb 1 23:23:59] NOTICE[13753]:
Re: [asterisk-users] asterisk 1.8 fax woes
This does sound like something that should stay on Asterisk-users. On Sat, Nov 13, 2010 at 3:36 AM, Jeremy Kister asterisk...@jeremykister.com wrote: I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. So you made sure to remove the res_fax.so module that was there from 1.6.2? Tried cleaning out the modules directory then installing just the 1.8 modules to be safe? All my custom modules (including swift thanks darren!) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip set debug peer vgw1 (vgw1 is my cisco 1760 ata) http://jeremy.kister.net/tmp/fax/console.txt http://jeremy.kister.net/tmp/fax/messages.txt http://jeremy.kister.net/tmp/fax/sip.txt I've tried using the packaged app_fax_spandsp and also Digium's app_fax_digum for 1.8.0-rc1 -- no difference in behavior. Anyone have any ideas how I can get this fixed? Have you tried doing tests where you send all calls straight into ReceiveFax and disable faxdetect? That may help track down where the problem is at least. You can put a noop before the call to receivefax if you'd like, but keep it simple and don't do anything else for this part of the test. If you've got a paid for Fax license (as opposed to Free Fax) then you can also contact Digium Support. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TELUS British Columbia PRI Settings
[trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0,esf,b8zs If you're only using one span, is there a reason you are using trunkgroups? I believe those only get used for NFAS and GR-303 #include /etc/asterisk/dahdi-channels.conf Do you have anything defined in this file? Since it comes at the top, any changes you make below it won't affect anything defined in that file. bchannel = 1-12 dchannel = 24 I didn't think bchannel and dchannel were valid for chan_dahdi.conf. Don't those only exist in system.conf? I believe you only declare 'channel' for the b-channels in chan_dahdi.conf. http://svn.asterisk.org/svn/asterisk/branches/1.6.2/configs/chan_dahdi.conf.sample -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TELUS British Columbia PRI Settings
Oh, and that isn't how a spanmap looks either. It looks like you have mixed some stuff from system.conf and chan_dahdi.conf here. My guess is your system.conf is configured at least mostly right, and that is why everything goes green. http://svn.asterisk.org/svn/dahdi/tools/branches/2.3/system.conf.sample On Sat, Aug 28, 2010 at 8:35 AM, Charles Moye cha...@gmail.com wrote: [trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0,esf,b8zs If you're only using one span, is there a reason you are using trunkgroups? I believe those only get used for NFAS and GR-303 #include /etc/asterisk/dahdi-channels.conf Do you have anything defined in this file? Since it comes at the top, any changes you make below it won't affect anything defined in that file. bchannel = 1-12 dchannel = 24 I didn't think bchannel and dchannel were valid for chan_dahdi.conf. Don't those only exist in system.conf? I believe you only declare 'channel' for the b-channels in chan_dahdi.conf. http://svn.asterisk.org/svn/asterisk/branches/1.6.2/configs/chan_dahdi.conf.sample -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid
Hi,
I have two asterisk servers with the same version of 1.4.29.1.
The first server named it as MYE1. MYE1 is an incoming server that can
accept incoming calls from PSTN(ZAP E1).
The second server is a pbx functions server and named it as MYPBX(SIP).
The sip.conf of MYE1 likes below:
[MYPBX]
type=peer
host=mypbx.abc.com
nat=no
disallow=all
allow=g729
canreinvite=yes
qualify=no
context=default
insecure=port,invite
The sip.conf of MYPBX likes below:
[MYE1]
type=peer
host=mye1.abc.com
nat=no
disallow=all
allow=g729
canreinvite=yes
qualify=no
context=did
insecure=port,invite
The call flow is
1. Mobile with disable callerid(+886-912-345678) make a call to DIDs on the
E1 (for example: +886-922-66 and enters MYE1 system. But my telecomm
provider helps me to solve the callerid and make it enable. So that, I can
find callerid of Mobile from MYE1.
2. MYE1 accept this call and dial it to MYPBX. In this moment, I can find
the fllowing message on the CLI of MYE1.
In Another word, the Caller ID is correct here.
-- Accepting call from '912345678' to '092266' on channel 0/22, span
4
-- Executing [0922666...@default:1] Set(DAHDI/94-1,
CDR(userfield)=0922E1) in new stack
-- Executing [0922666...@default:2] Set(DAHDI/94-1,
CALLERID(num)=912345678) in new stack
-- Executing [0922666...@default:3] Set(DAHDI/94-1,
CALLERID(num)=912345678) in new stack
-- Executing [0922666...@default:4] NoOp(DAHDI/94-1, CID num:
[986230883]) in new stack
-- Executing [0922666...@default:5] Dial(DAHDI/94-1, SIP/
mypbx.abc.com/092266) in new stack
-- Called mypbx.abc.com/092266
-- SIP/mypbx.abc.com-2551 is ringing
extensions.conf
exten = 092266,1,Set(CDR(userfield)=0922E1)
exten = 092266,n,NoOp(CID num: [${CALLERID(num)}])
exten = 092266,n,Set(CALLERID(num)=${CALLERID(num)})
exten = 092266,n,NoOp(CID num: [${CALLERID(num)}])
exten = 092266,n,Dial(SIP/mypbx.abc.com/${EXTEN})
exten = 092266,n,Hangup
3. But the strange thing is MYPBX. I use the function NoOp to find the
callerid that call from MYE1.
-- Executing [0922666...@did:1] NoOp(SIP/MYE1-0185, CID Num:
Anonymous) in new stack
-- Executing [0922666...@did:2] Hangup
extensions.conf
exten = _X.,1,NoOp(CID Num: ${CALLERID(number)})
exten = _X.,1,Hangup
4. I got the ngrep message from MYPBX.
U 210.200.XXX.XX:5060 - 61.65.XX.XX:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
61.65.XX.XX:5060;branch=z9hG4bK276d72eb;received=61.65.XX.XX;rport=5060.
From: Anonymous sip:anonym...@anonymous.invalid;tag=as2b63fbb6.
To: sip:0922666...@mypbx.abc.com sip%3a0922666...@mypbx.abc.com.
Call-ID: 1a9bc32e05eeb8b27bfadfba35c09...@61.65.xx.xx.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
Supported: replaces.
Contact: sip:0922666...@210.200.xxx.xx.
Content-Length: 0.
.
U 210.200.XXX.XX:5060 - 61.65.XX.XX:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP
61.65.XX.XX:5060;branch=z9hG4bK276d72eb;received=61.65.XX.XX;rport=5060.
From: Anonymous sip:anonym...@anonymous.invalid;tag=as2b63fbb6.
To: sip:0922666...@xm1.gvlink.net sip%3a0922666...@xm1.gvlink.net
;tag=as66351139.
Call-ID: 1a9bc32e05eeb8b27bfadfba35c09...@61.65.xx.xx.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
Supported: replaces.
5. My questions are:
A. Why can't I receive the CALLERID from MYPBX(the secondary server)? I
am sure I use Set(CALLERID(num) for it.
B. Why does the CALLERID that sends from MYE1 become as Anonymous? How
can I fix it with the correct orginal callerid(912345678)?
C. Why does my FROM message become as Anonymous
sip:anonym...@anonymous.invalid instead of 912345...@mye1.abc.com ?
If you have any suggestions, please let me know. Thank you very much.
--
Best Regards
Charles
--
_
-- 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] Twitter is Suing me!!!
Good Luck, I went though this with Yahoo in the early 2000s. Their basic argument is that their mark is included in your mark and they want your domain. They are domain bullies. I went ahead and purchased your app because it sounded pretty cool. I wish the best for you. On Tue, Aug 11, 2009 at 10:28 PM, Dean Collins d...@cognation.net wrote: This isn’t asterisk related but I figure several developers on this list have built apps for Twitter (or other 3rd party API’s). Just found out a few hours ago I’m being sued by Twitter Feel free to tweet this link ( www.MyTwitterButler.com/I’m_Being_Suedhttp://www.mytwitterbutler.com/I'm_Being_Sued) or forward on the link to any journalists you know. If you are on dig here’s a dig link. http://digg.com/software/My_Twitter_Butler_I_m_Being_Sued Regards, Dean Collins d...@mytwitterbutler.comd...@mytwitterbutler.com?subject=i'm%20being%20Sued +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -- *From:* Dean Collins *Sent:* Tuesday, August 11, 2009 10:37 PM *Subject:* Twitter is Suing me!!! So not only are Twitter fighting battles with Russian hackers they are now fighting their own third party API developer community !! I received this email 30 minutes ago stating that Twitter is suing me?? Basically they feel that my application - www.MyTwitterButler.comhttp://www.mytwitterbutler.com/does the following. * * *1/ That anyone using the API to auto follow people are breaching the TOS??* * * *2/ That no one can use the word “Twitter” in their domain* *3/ That somehow people might be confused my application is related to twitter even though every page is labeled * *“**Copyright 2009 © My Twitter Butler - Not related in anyway to Twitter Inc, if I owned Twitter would i be spending my time building this app??* Is this the end for Twitter 3rd party developers? Have they forgotten that it was people like me who saw a need and built an application using the publicly defined Twitter API to add value to the Twitter ecosystem? I have asked Twitters lawyers for a conference call tomorrow to clear up ‘WHY’ they feel anyone using the twitter API to auto follow people is an illegal act and will be looking forward to their answers about ‘WHY’ the twitter API was built in the first place if they want to sue people for using it. www.MyTwitterButler.com/I’m_Being_Suedhttp://www.mytwitterbutler.com/I'm_Being_Sued Regards, Dean Collins d...@mytwitterbutler.com +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To: Field
Ah that is brilliant, thanks a lot.
Charles
On Mon, Jun 1, 2009 at 9:35 AM, Administrator TOOTAI ad...@tootai.netwrote:
Hi
Charles Solar a écrit :
Hi guys, I am new here but I have a quick question.
I have an incoming trunk that sends me calls from various usernames I
have
with them. Only trouble is they send invites as s...@my.ip.addr, not as
the
username I have with them. So I cannot match extensions like I would
want
to.
Here is a sample invite
INVITE sip:s...@my.ip.ad.dr SIP/2.0
Record-Route: sip:0.0.0.0;lr=on;ftag=as29ffee59
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK7238.25c90fc7.0
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK1c709971;rport=5060
From: sip:999...@host.ip.addr;tag=as29ffee59
To: sip:myusern...@mysipprovider.netsip%3amyusern...@mysipprovider.net
sip%3amyusern...@mysipprovider.net sip%253amyusern...@mysipprovider.net
Contact: sip:999...@host.ip.addr
Call-ID: 6a379af207d78b3b5f2e8c6c55e64009
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Fri, 29 May 2009 04:12:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 377
the only distinction between a call to username1 and username2 is in the
To:
field, but I cannot find something to route the call based on the To
caller
id.
I think the dialednumber variable would be close to what I want, but
apparently that is broken so I am unsure what to do.
[macro-setDialednumberFromSipHeader]
;
; We extract the DIALEDNUMBER from SIP header
; which is of the form sip:callednum...@ourasteriskipaddress
exten = s,1,Set(__DIALEDNUMBER=${SIP_HEADER(TO):5})
exten = s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,@,1)})
exten = s,n,GotoIf($[${DIALEDNUMBER:0:1} != +]?numberIsOK)
exten = s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,+,2)})
exten = s,n(numberIsOK),NoOp()
exten = s,n,Set(CDR(dest)=${DIALEDNUMBER})
done ;-)
___
-- 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] To: Field
Again, another brilliant solution that I was unaware of :D Thanks so much On Mon, Jun 1, 2009 at 10:24 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I have an incoming trunk that sends me calls from various usernames I have with them. Only trouble is they send invites as s...@my.ip.addr, not as the username I have with them You need to adjust your register = statement with them: Add /username to the end of it, then calls won't arrive at 's' but at username. Philipp ___ -- 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] To: Field
Hi guys, I am new here but I have a quick question. I have an incoming trunk that sends me calls from various usernames I have with them. Only trouble is they send invites as s...@my.ip.addr, not as the username I have with them. So I cannot match extensions like I would want to. Here is a sample invite INVITE sip:s...@my.ip.ad.dr SIP/2.0 Record-Route: sip:0.0.0.0;lr=on;ftag=as29ffee59 Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK7238.25c90fc7.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK1c709971;rport=5060 From: sip:999...@host.ip.addr;tag=as29ffee59 To: sip:myusern...@mysipprovider.net sip%3amyusern...@mysipprovider.net Contact: sip:999...@host.ip.addr Call-ID: 6a379af207d78b3b5f2e8c6c55e64009 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Fri, 29 May 2009 04:12:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 377 the only distinction between a call to username1 and username2 is in the To: field, but I cannot find something to route the call based on the To caller id. I think the dialednumber variable would be close to what I want, but apparently that is broken so I am unsure what to do. Thanks for any pointers ___ -- 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] Asterisk 1.6 pbx_lua not creating any contexts
To follow up -- pbx_lua from trunk works as advertised when backported to 1.6. pbx_lua from asterisk 1.6 seems hopelessly broken, and I've given up on trying to persuade it to work. ___ -- 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 1.6 pbx_lua not creating any contexts
Howdy, all. I'm trying to use pbx_lua as included in Asterisk 1.6 -- but while it correctly reports an error on startup (but not reload!) if extensions.lua does not exist, it doesn't appear to actually create any contexts. I'm testing in a very minimal configuration with autoload turned off; module show shows only chan_sip, pbx_lua, and app_dial. dialplan show calls only 'app_dial_gosub_virtual_context', created by app_dial, and 'parkedcalls', created by 'features'; no contexts defined in extensions.lua are visible. This is true even when using verbatim the extensions.lua.sample included in SVN (trunk as of r144523). pbx_config, if enabled, works normally. Where should I start in diagnosing this issue? 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] DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away!
Since I play a 70 balance druid on WoW I thought it was something else. On Tue, Sep 9, 2008 at 2:56 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 09 September 2008 16:36:37 Dean Collins wrote: There have been at least 4 announcements with dates etc, this is really just the last chance reminder email. It's the first I've seen of it. In any case, if this was the last in a series of reminders, I'm puzzled why the email started with the phrase proud to announce. That would either indicate that this was intended as the first announcement, or the person posting the reminder failed to review the text. Neither is a very comforting thought. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf templates and realtime
I currently have my phones setup in the sip.conf file. I use templates to describe the specific settings to each phone type. For instance in sip.conf, I have: [generic_phone](!) ... ... [polycom501](!,generic_phone) ... ... [grandstream](!,generic_phone) ... ... ;begin subscribers [200](polycom501) ... ... [201](grandstream) ... ... I am using asterisk 1.4.21.2 I would like to move my sip users to realtime, so my questions are: 1) Can I continue to use the templates from sip.conf and the template settings get passed to realtime and if so, how? In the comments in the sip.conf file where it shows the User config options ant Peer configuration, on the peer side it shows a template field, which seems to indicate to me that this can be done. 2) If this is not the purpose of the template field, what is it's purpose? I can not seem to find it documented anywhere. Note: I do not have any problems getting realtime to work, as long as I put every field that is needed (or required) in each record, but I think life would be easier if I could leave my templates (that rarely change) in the sip.conf file and put the bare necessities in realtime (users that change all the time). Thanks, Charles Wadsworth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with 3-way conferenicing
Hi,
I think the important error message is jumping out of macro
'nway-conf-start' not ast_bridge_call.
It is because it is not allow to jump to another context when you use macro.
Best regards,
Charles
2007/4/23 Manu Mehta [EMAIL PROTECTED]:
Hi,
I am trying to achieve 3-way conferencing taking hint from wiki link
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
Here is the scenario:
1. user ua1 calls user ca1
2. ua1 then presses the feature code *0 to redirect ca1 to conference
room 300
3. ua1 then dials the user 33
4. user ua1 and 33 are connected
5. Now when ua1 presses the feature code ** to redirect user 33 to
same conference room 300, there is error thrown on Asterisk console that
res_features.c:1415 ast_bridge_call: Bridge failed on channels
SIP/ua1-ac750040 and AsyncGoto/Local/[EMAIL PROTECTED],1ZOMBIE
Here is my dial plan:
*[manu]*
exten = ca1,1,Dial(SIP/ca1,,wWtTkKrR)
*[nway-conf]*
exten = _.,1,Answer
exten = _.,n,Set(CONFNO=${EXTEN})
exten = _.,n,Set(MEETME_EXIT_CONTEXT=nway-conf-invite)
exten = _.,n,Set(DYNAMIC_FEATURES=)
exten = _.,n,MeetMe(${CONFNO},pdMX)
exten = _.,n,Hangup
*[nway-conf-invite]*
exten = 0,1,Read(DEST,dial,,i)
exten = 0,n,Set(DYNAMIC_FEATURES=nway-conf-inv#nway-conf-noinv)
exten = 0,n,Dial(Local/[EMAIL PROTECTED],,g)
exten = 0,n,Set(DYNAMIC_FEATURES=)
exten = 0,n,Goto(nway-conf,${CONFNO},1)
exten = i,1,Goto(nway-conf,${CONFNO},1)
*[nway-conf-dest] *
exten = _.,1,Dial(SIP/${EXTEN})
*[macro-nway-conf-start] *
exten = s,1,Set(CONFNO=300)
exten = s,n,ChannelRedirect(${BRIDGEPEER},nway-conf,${CONFNO},1)
exten = s,n,Read(DEST,dial,,i)
exten = s,n,Set(DYNAMIC_FEATURES=nway-conf-inv#nway-conf-noinv)
exten = s,n,Dial(Local/[EMAIL PROTECTED],,g)
exten = s,n,Set(DYNAMIC_FEATURES=)
exten = s,n,Goto(nway-conf,${CONFNO},1)
*[macro-nway-conf-ok] *
exten = s,1,ChannelRedirect(${BRIDGEPEER},nway-conf,${CONFNO},1)
The application map defined in features.conf is:
*[applicationmap] *
nway-conf-start = *0,self/caller,Macro,nway-conf-start
nway-conf-inv = **,self/caller,Macro,nway-conf-ok
nway-conf-noinv = *9,self/caller,Macro,nway-conf-notok
*The output logs on Asterisk console:*
localhost*CLI
localhost*CLI
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/ua1-ac750040,
SIP/ca1||wWtTkKr) in
new stack
-- Called ca1
-- SIP/ca1-ab110040 is ringing
-- SIP/ca1-ab110040 answered SIP/ua1-ac750040
[Apr 19 16:14:12] WARNING[22989]: rtp.c:874 ast_rtcp_read: RTCP Read too
short
-- Feature Found: nway-conf-start exten: nway-conf-start
-- Executing [EMAIL PROTECTED]:1] Set(SIP/ua1-ac750040,
CONFNO=300) in new stack
-- Executing [EMAIL PROTECTED]:2]
ChannelRedirect(SIP/ua1-ac750040, SIP/ca1-ab110040|nway-conf|300|1) in
new stack
-- Executing [EMAIL PROTECTED]:3] Read(SIP/ua1-ac750040,
DEST|dial||i) in new stack
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/ca1-ab110040, ) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/ca1-ab110040, CONFNO=300) in
new stack
-- Executing [EMAIL PROTECTED]:3] Set(SIP/ca1-ab110040,
MEETME_EXIT_CONTEXT=nway-conf-invite) in new stack
-- Executing [EMAIL PROTECTED]:4] Set(SIP/ca1-ab110040,
DYNAMIC_FEATURES=) in new stack
-- Executing [EMAIL PROTECTED]:5] MeetMe(SIP/ca1-ab110040, 300|pdMX) in
new stack
-- Created MeetMe conference 1023 for conference '300'
-- Playing 'conf-onlyperson' (language 'en')
[Apr 19 16:14:15] WARNING[22995]: rtp.c:874 ast_rtcp_read: RTCP Read too
short
-- Started music on hold, class 'default', on SIP/ca1-ab110040
-- User entered '33'
-- Executing [EMAIL PROTECTED]:4] Set(SIP/ua1-ac750040,
DYNAMIC_FEATURES=nway-conf-inv#nway-conf-noinv) in new stack
-- Executing [EMAIL PROTECTED]:5] Dial(SIP/ua1-ac750040,
Local/[EMAIL PROTECTED]||g) in new stack
-- Called [EMAIL PROTECTED]
-- Executing [EMAIL PROTECTED]:1] Dial(Local/[EMAIL PROTECTED],2,
SIP/33) in new stack
-- Called 33
[Apr 19 16:14:18] WARNING[22995]: rtp.c:874 ast_rtcp_read: RTCP Read too
short
-- SIP/33-a8ff0040 is ringing
-- Local/[EMAIL PROTECTED],1 is ringing
-- SIP/33-a8ff0040 is ringing
-- SIP/33-a8ff0040 is ringing
-- SIP/33-a8ff0040 answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 stopped sounds
-- Local/[EMAIL PROTECTED],1 answered SIP/ua1-ac750040
[Apr 19 16:14:21] WARNING[22995]: rtp.c:874 ast_rtcp_read: RTCP Read too
short
[Apr 19 16:14:24] WARNING[22995]: rtp.c:874 ast_rtcp_read: RTCP Read too
short
-- Feature Found: nway-conf-inv exten: nway-conf-inv
-- Executing [EMAIL PROTECTED]:1] Set(SIP/ua1-ac750040,
CONFNO=300) in new stack
-- *Executing [EMAIL PROTECTED]:2] ChannelRedirect(SIP/ua1-ac750040,
Local/[EMAIL PROTECTED],1|nway-conf|300|1) in new stack *
*[Apr 19 16:14:25] WARNING[22989]: res_features.c:1415 ast_bridge_call:
Bridge failed on channels SIP/ua1-ac750040 and
AsyncGoto/Local/[EMAIL PROTECTED],1ZOMBIE *
-- Executing [EMAIL PROTECTED]:6] Set(SIP/ua1-ac750040,
DYNAMIC_FEATURES=) in new stack
-- Executing [EMAIL PROTECTED]:7
Re: [asterisk-users] freecall.com - has anybody tried it?
I used the same service and bought EURO $10 from www.freecall.com. But I can't make calls to China at all. I can use only in Taiwan. There is contact phone number but no one answer the phone. And nobody give me any reponse after I write the feeback from its website. 2007/2/26, Ira [EMAIL PROTECTED]: At 09:10 AM 2/25/2007, you wrote: I don't have any qualified Windows box to get an account and try it. Can anybody comment on setup and or call quality? I've been using it for 6 or 8 months for my calls to New Zeland and Australia. It's been perfectly acceptable but the people I call know it's free so they put up with the occasional issues or I just call back. I tried using it for domestic US but I can't set callerid and the servers seem to be far away from Los Angeles so I use domestic services for domestic calls. I recommend it to friends who need to make overseas calls because it seems to be the best service I've run across for that purpose. FWIW, the free calls only last 90 days after you deposit the 10 euros and then you use that up and get another 90 days free or that's how it seems to work. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ -- 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] Test
Test Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- 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] Astersik Transcoder support
Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- 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] Good, affordable IAX hardphones?
We use: http://www.ngnsky.com/product_info.php?cPath=21products_id=50 when we have the remote extension blues. It works quite well for us and the phone isn't that bad. On 10/19/07, Vincent [EMAIL PROTECTED] wrote: Hi SIP is such a pain to use when NAT is involved that I'm willing to buy an IAX hardphone for someone who works remotely over the Net and needs to get calls from our Asterisk server, itself behind a NAT. Are there good, affordable IAX phones you would recommend? Thank you. ___ --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] Adtran feature codes, extensions
Greetings, We have An Adtran 616 Total Access device talking to a colocated Asterisk machine over MGCP. Calls placed to the phones connected to the Adtran go through as do outgoing calls from the phone (prefixed by 9), but feature access codes (*97 for voicemail, for example) and extension-to-extension calls don't work. As soon as the first digit is pressed, the user hears a busy signal. I confesss to not knowing much about how MGCP works, but I can't seem to find any kind of digit map in the Adtran so is Asterisk the one listening for but not acknowledging these digits? What do I have to do to make these work? Any help appreciated. Thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Audio with Gtalk
Dear Michael, I got the same problem for a long time, but noboday give me some tips. Do you solve it? Best regards, Charles 2007/4/1, Michael Zoller [EMAIL PROTECTED]: I configured my * with the instructions found here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk to work with gtalk. The Phone rings and connects - but no audio! I am using a self-compiled asterisk 1.4.2 There is a lot of output on the CLI but I can't make sense of it. Perhaps somebody can help? Michael Output from the CLI: JABBER: gtalk_account OUTGOING: iq type='result' from='[EMAIL PROTECTED]/TalkB0AA717E' to='[EMAIL PROTECTED]/Talk.v10402D9EB78' id='317'/ atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=d type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=e type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ [Apr 1 09:50:28] NOTICE[20781]: chan_gtalk.c:1333 gtalk_indicate: Don't know how to indicate condition '-1' JABBER: gtalk_account OUTGOING: iq type='set' to='[EMAIL PROTECTED]/Talk.v10402D9EB78' from='[EMAIL PROTECTED]/TalkB0AA717E' id='f'session xmlns='http://www.google.com/session' type='accept' initiator='[EMAIL PROTECTED]/Talk.v10402D9EB78' id='2926563865'description xmlns='http://www.google.com/session/phone' xml:lang='en'payload-type id='0' name='PCMU' clockrate='8000' bitrate='64000'/payload-type id='100' name='EG711U' clockrate='8000' bitrate='64000'/payload-type id='106' name='telephone-event' clockrate='8000'//descriptiontransport xmlns='http://www.google.com/transport/p2p'//session/iq atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=f type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ atlas*CLI JABBER: gtalk_account OUTGOING: iq type='set' from='[EMAIL PROTECTED]/TalkB0AA717E' to='[EMAIL PROTECTED]/Talk.v10402D9EB78' id='g'session type='terminate' id='2926563865' initiator='[EMAIL PROTECTED]/Talk.v10402D9EB78' xmlns='http://www.google.com/session'//iq atlas*CLI JABBER: gtalk_account INCOMING: iq to=[EMAIL PROTECTED]/TalkB0AA717E id=g type=result from=[EMAIL PROTECTED]/Talk.v10402D9EB78/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NO ANSWER, When openser make an oubound SIP call to my asterisk
Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip debug in my asterisk CLI. Does anybody kind to help me to solve it or give me some tips please? Best regards, Charles # my asterisk CLI [EMAIL PROTECTED] ~]# asterisk -rvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Binding iaxusers to mysql/asterisk/iaxfriends == Binding iaxpeers to mysql/asterisk/iaxfriends == Binding queues to mysql/asterisk/queue_table == Binding queue_members to mysql/asterisk/queue_member_table Asterisk 1.2.16, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.16 currently running on asterisk (pid = 26311) Verbosity is at least 14 -- Remote UNIX connection asterisk*CLI sip debug SIP Debugging re-enabled asterisk*CLI # my command running on asterisk machine: ngrep -t -W byline -d any port 5060 interface: any filter: (ip) and ( port 5060 ) # U 2007/05/17 13:31:35.908163 my.openser.ip.addr:5060 - my.asterisk.ip.addr :5060 INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. Record-Route: sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx ;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 sip:[EMAIL PROTECTED];tag=3840196923. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]:57536. Call-ID: [EMAIL PROTECTED] CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr. t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:36.325713 my.openser.ip.addr:5060 - my.asterisk.ip.addr :5060 INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. Record-Route: sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx ;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 sip:[EMAIL PROTECTED];tag=3840196923. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]:57536. Call-ID: [EMAIL PROTECTED] CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr. t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:37.325722 my.openser.ip.addr:5060 - my.asterisk.ip.addr :5060 INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. Record-Route: sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx ;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 sip:[EMAIL PROTECTED];tag=3840196923. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]:57536. Call-ID: [EMAIL PROTECTED] CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr. t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:39.325425 my.openser.ip.addr:5060 - my.asterisk.ip.addr :5060 INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. Record-Route: sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP
Re: [asterisk-users] Hardware requirements question
On Saturday 14 April 2007 00:52, [EMAIL PROTECTED] wrote: Can you tell me if this sounds sane? We are planning on using a Dell 933Mhz dual CPU server, with 1GB of ram for our Trixbox setup. We will have 7-10 internal phones, and maybe 3-4 max outbound connections at a time. We will have some type of menu system for inbound callers. At this point I'm planning on connecting to a SIP provider over the internet for service. Do you think the hardware is adequate? If there's a chance its not enough horsepower I want to find a different server. I'm not an expert, but I'd say that this is pretty much spot-on for what you're trying to do. We've deployed systems before with twice the number of extensions and half the horsepower with no problems. -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play blank sound while VM recording?
On Tuesday 03 April 2007 20:09, [EMAIL PROTECTED] wrote: Charles Ulrich wrote: I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This is fine and probably should be the expected behavior, except that after 30 seconds to a minute of not seeing any RTP traffic coming from the PBX, the trunk appears to make the faulty assumption that the PBX is gone and hangs up the call. Maybe this is what you need?: ;rtpkeepalive=secs; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero) (in sip.conf, [general] section) Regards, Philipp That was exactly what I needed, thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Play blank sound while VM recording?
Greetings, (Apologies if this is an FAQ, but I've Googled for hours and haven't come up with anything yet.) I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This is fine and probably should be the expected behavior, except that after 30 seconds to a minute of not seeing any RTP traffic coming from the PBX, the trunk appears to make the faulty assumption that the PBX is gone and hangs up the call. I've called the trunk provider and they said two things. 1) This is indeed what their trunk was programmed to do. 2) No, they won't change it. We're working on switching the customer to a trunk provider with a bit more clue, but in the meantime, how can I have Asterisk play an empty sound file while the caller is leaving a voicemail message just to keep the RTP traffic flowing? This installation of Asterisk was designed by someone else and I have limited personal experience with Asterisk configuration files, so an example would be appreciated if possible. Thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GTalk/Jabber passing audio in 1.4.1!
Dear Lewis, Can you please post you gtalk.conf and jabber.conf for me? I also make it under Fedora Core 6. But I got no audio at all. I use X-Lite as SIP client (under NAT). 2007/3/7, Ronald Lewis [EMAIL PROTECTED]: I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got two-way audio between Google Talk and Asterisk! This IS an exciting moment today in VoIP! This is just GREAT! - Ronald Lewis http://ronaldlewis.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call was hangup when LIMIT_WARNING_FILE was playing
Dear All,
I tried to use 'L' option on my dial command.
I set the x to 65000(65 seconds), y to 6(60 seconds), z to
3(30 seconds).
The max calltime should be 65 seconds, and it will play beep.gsm at
60 seconds left. And repeat the beep at 30 seconds left.
But the call will be hangup by system at 60 seconds left.
In another word, when it plays warning file, the call will be hangup.
The answeredtime is only 5 seconds.
Can anybody give me an idea for it?
*** extensions.conf ***
[default]
exten= _+[1-9].,1,SetCallerID()
exten= _+[1-9].,2,Set(LIMIT_WARNING_FILE=beep)
exten= _+[1-9].,3,Set(LIMIT_TIMEOUT_FILE=beep)
exten= _+[1-9].,4,Dial(zap/g1/002${EXTEN:1}|60|L(65000:6:3))
exten= _+[1-9].,105,Hangup
Log from CLI
***
-- Seeding '24012100' at 61.217.XXX.XXX:8625 for 60
-- Accepting AUTHENTICATED call from 61.217.XXX.XXX:
requested format = ilbc,
requested prefs = (),
actual format = ilbc,
host prefs = (ilbc),
priority = mine
-- Executing SetCallerID(IAX2/24012100-2, ) in new stack
-- Executing Set(IAX2/24012100-2, LIMIT_WARNING_FILE=beep) in new stack
-- Executing Set(IAX2/24012100-2, LIMIT_TIMEOUT_FILE=beep) in new stack
-- Executing Dial(IAX2/24012100-2,
zap/g1/0028621|60|L(65000:6:3)) in new stack
-- Limit Data for this call:
-- - timelimit = 65000
-- - play_warning = 6
-- - play_to_caller= yes
-- - play_to_callee= no
-- - warning_freq = 3
-- - start_sound = UNDEF
-- - warning_sound = beep
-- - end_sound = beep
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/0028621
-- Zap/29-1 is proceeding passing it to IAX2/24012100-2
-- Zap/29-1 is ringing
-- Zap/29-1 answered IAX2/24012100-2
-- Hungup 'Zap/29-1'
== Spawn extension (default, +8621, 4) exited non-zero on
'IAX2/24012100-2'
-- Hungup 'IAX2/24012100-2'--
Best Regards
Charles
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?
Dear Phil,
Thank you for your reply.
I have changed by extensions.conf as below.
And I also put my debug information for your reference.
It is a strange behavior. I got exited non-zero in it when I use ZAP channel.
If I use my SIP trunking gateway(outside), I got the return value is zero.
** extensions.conf **
exten= _00[1-9].,1,Dial(zap/g1/${EXTEN})
exten= _00[1-9].,h,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
exten= _00[1-9].,102,Hangup
***
myserver*CLI agi debug
AGI Debugging Enabled
-- Seeding '24012100' at 61.217.xxx.xxx:8400 for 60
-- Accepting AUTHENTICATED call from 61.217.xxx.xxx:
requested format = ilbc,
requested prefs = (),
actual format = ilbc,
host prefs = (ilbc),
priority = mine
-- Executing Dial(IAX2/24012100-1, zap/g1/008621) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/0028621
-- Zap/29-1 is proceeding passing it to IAX2/24012100-1
-- Zap/29-1 is ringing
-- Zap/29-1 answered IAX2/24012100-1
-- Hungup 'Zap/29-1'
== Spawn extension (default, 008621, 1) exited non-zero on
'IAX2/24012100-1'
-- Hungup 'IAX2/24012100-1'
2007/2/21, Phil Reynolds [EMAIL PROTECTED]:
Quoting Charles Wang [EMAIL PROTECTED]:
Dear all,
I tried to make a call with extensions.conf.
exten= _00[1-9].,1,Dial(zap/g1/${EXTEN})
exten= _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
exten= _00[1-9].,102,Hangup
But the 2 and 102 will not be executed.
So I can get the correct answered time via 2.
Is any idea about it?
The Dial() exits when the call is finished - then control passes to
the h extension if present.
Therefore, I think you need to put the NoOp in the h extension. It
only continues at 2 if the Dial() times out.
Not sure but that's how I understand it.
--
Phil Reynolds
o mail: [EMAIL PROTECTED]
|L_ \ / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
--
Best Regards
Charles
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?
Dear Phil,
The extension 'h' was a great idea although I still got the error
exited non-zero.
Thank you for your help.
Best regards,
Charles
2007/2/21, Phil Reynolds [EMAIL PROTECTED]:
Quoting Charles Wang [EMAIL PROTECTED]:
Dear Phil,
Thank you for your reply.
I have changed by extensions.conf as below.
And I also put my debug information for your reference.
It is a strange behavior. I got exited non-zero in it when I use ZAP channel.
If I use my SIP trunking gateway(outside), I got the return value is zero.
** extensions.conf **
exten= _00[1-9].,1,Dial(zap/g1/${EXTEN})
exten= _00[1-9].,h,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
Still wrong... exten = h,1,NoOp...
exten= _00[1-9].,102,Hangup
This line is superfluous.
--
Phil Reynolds
o mail: [EMAIL PROTECTED]
|L_ \ / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
--
Best Regards
Charles
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't get ANSWEREDTIME after hangup using ZAP
Dear all, I tried to make a call with PHP AGI. $rc = execute_agi(EXEC DIAL ZAP/g1/$myphonenumber|60|rhHL( . ($max_total_seconds * 1000) . :6:3) ); $rc = execute_agi(GET VARIABLE ANSWEREDTIME ); And I can't get the answered time after caller hangup in this method. But if I use a SIP channel as below: $rc = execute_agi(EXEC DIAL SIP/$mysiptrunk/$myphonenumber|60|rhHL( . ($max_total_seconds * 1000) . :6:3) ); $rc = execute_agi(GET VARIABLE ANSWEREDTIME ); I can get the correct answered time. Is any idea about it? Is it the problem of my ZAP channel's configuration? -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?
Dear all,
I tried to make a call with extensions.conf.
exten= _00[1-9].,1,Dial(zap/g1/${EXTEN})
exten= _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
exten= _00[1-9].,102,Hangup
But the 2 and 102 will not be executed.
So I can get the correct answered time via 2.
Is any idea about it?
Is it the problem of my ZAP channel's configuration?
My zapata.conf is as below:
[channels]
language=en
context=default
busydetect=no
callprogress=no
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
overlapdial=yes
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
channel = 1-15,17-31
--
Best Regards
Charles
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1,4 and google talk
I also got the same problem on my Fedora Core 6, too. 2006/11/7, Mani Sridhar [EMAIL PROTECTED]: hi fellow asterisk enthusiasts, i've configured jabber.conf and gtalk.conf as descibed on voip-info.org (http://www.voip-info.org/wiki/view/Asterisk+Google+Talk). i see these messages on the CLI now, and i haven't been able to get Asterisk-Gtalk connectivity to work. *CLI [Nov 3 22:17:01] WARNING[30878]: res_jabber.c:1504 aji_recv_loop: JABBER: socket read error *CLI JABBER: gtalk_account OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='gmail.com' version='1.0' *CLI JABBER: gtalk_account INCOMING: ?xml version=1.0 encoding=UTF-8?stream:stream from=gmail.com id=D428120132AB91B7 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client [Nov 3 22:17:01] ERROR[30878]: res_jabber.c:482 aji_act_hook: gnuTLS not installed. *CLI JABBER: gtalk_account INCOMING: stream:featuresstarttls xmlns=urn:ietf:params:xml:ns:xmpp-tls/mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanism/mechanisms/stream:features *CLI these messages just keep appearing every 20s. gnuTLS is installed, so the error message gnuTLS not installed does not make sense to me. i checked config.log after running ./configure while building asterisk, and i can see that the check for gcc -lgnutls passed. [EMAIL PROTECTED] asterisk]# rpm -qi gnutls Name : gnutls Relocations: (not relocatable) Version : 1.0.25 Vendor: Red Hat, Inc. Release : 2.FC4 Build Date: Fri 10 Feb 2006 02:51:42 PM PST Install Date: Tue 31 Oct 2006 03:21:16 PM PST Build Host: hs20-bc1-7.build.redhat.com Group : System Environment/Libraries Source RPM: gnutls-1.0.25-2.FC4.src.rpm Size : 664600 License: LGPL Signature : DSA/SHA1, Fri 10 Feb 2006 05:10:47 PM PST, Key ID b44269d04f2a6fd2 Packager : Red Hat, Inc. http://bugzilla.redhat.com/bugzilla URL : http://www.gnutls.org/ Summary : A TLS implementation. Description : The GNU TLS library implements TLS. Someone needs to fix this description. [EMAIL PROTECTED] asterisk]# [EMAIL PROTECTED] asterisk]# ls -la /usr/lib/*gnutls* lrwxrwxrwx 1 root root 26 Oct 31 15:21 /usr/lib/libgnutls-extra.so.11 - libgnutls-extra.so.11.1.25 -rwxr-xr-x 1 root root 163832 Feb 10 2006 /usr/lib/libgnutls-extra.so.11.1.25 lrwxrwxrwx 1 root root 28 Oct 31 15:21 /usr/lib/libgnutls-openssl.so.11 - libgnutls-openssl.so.11.1.25 -rwxr-xr-x 1 root root 26756 Feb 10 2006 /usr/lib/libgnutls-openssl.so.11.1.25 lrwxrwxrwx 1 root root 20 Oct 31 15:22 /usr/lib/libgnutls.so - libgnutls.so.11.1.25 lrwxrwxrwx 1 root root 20 Oct 31 15:21 /usr/lib/libgnutls.so.11 - libgnutls.so.11.1.25 -rwxr-xr-x 1 root root 474012 Feb 10 2006 /usr/lib/libgnutls.so.11.1.25 [EMAIL PROTECTED] asterisk]# what can i check next? i'm pretty new (been working on asterisk for less than a month now) and i've been stuck at this point for a few days now. i'd really appreciate some pointers. thanks mani * Our reliance on access to a dialtone is now only slightly lesser than that on access to oxygen. _ Connect with your friends who use Yahoo! Messenger with Voice. Click! http://www.msnspecials.in/wlmyahoo/index.asp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 31, Issue 37
On Friday 09 February 2007 11:50, [EMAIL PROTECTED] wrote: Anyone got any experiences of good quality VoIP conferencing phones? I've used Polycom analogue units in the past, and I see that they have a SIP version (the IP4000) - but it is better/worse/as good as an analogue version? (ie. would I be better off with an analogue version into a TDM card or ATA?) Cheers, Gordon The quality of a conference phone is determined more by how it's designed and manufactured than whether it uses VoIP or analog. We've deployed a couple of IP4000s and they work great. The nice thing about them is that if you already have a bunch of SoundPoint IP phones, they require nothing special in regards to provisioning since they use the same firmware and configuration as the rest of the SoundPoint IP series. -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 31, Issue 29
On Thursday 08 February 2007 07:32, [EMAIL PROTECTED] wrote: Does anyone have any recommendations for a phone that has easy to understand/implement central provisioning? I've used CISCO 79XX phones, and they're great (but too expensive). I like Grandstream phones, but their provisioning sucks. What is everybody else using in large environments where individual config is not an option? We've been quite happy with Polycom SoundPoint IP phones. They are a tad pricey but they look nice, work well, and are extremely easy to deploy. They used to have poor NAT support but with the 2.0.x firmware, it's getting better. They can be provisioned with FTP, TFTP, HTTP, and HTTPS. The only complaint that I have with them is that their provisioning file format makes XML developers cry in sorrow. -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stress a server with a tool
The radvision's prolabs is your best choice for SIP or H.323. 2006/9/20, nik600 [EMAIL PROTECTED]: hi is there any software usable to simulate work on an asterisk server? I'm interested in it to evaluate the level of currently calls that a server can support ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Digium Zaptel volume issues
Is there a preferred card/capability that doesn't have the TDM400P limitations? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, July 14, 2006 3:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium Zaptel volume issues Charles K Green wrote: All, Anyone have any experience with the Digium TDM400P? We have a Digium TDM400P up and working with asterisk. We've fxotune'd the interface and pretty much eliminated all of the echo on the channel. Our latest issue is that all calls that run over the zap channels sound muffled and distant. I upped the rx gain and that helped some on the inbound voice but now we have some static creeping in. If I up the transmit gain, we start to get more echo so I'm really hesitant to do that too much. Lots of experience with it. I recently learned that a fixed loss is statically defined in the drivers for the card. The loss has been there since the card first came out and it is supposedly required to make the s/w EC function. The low audio issue is a common problem for all users, but it becomes far more noticeable for those that use the card on long pstn analog loops. In other words, the greater the pstn loss, the lower the audio level, and increasing rxgain/txgain cannot be used to compensate for it. No fix or workaround. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Zaptel volume issues
All, Anyone have any experience with the Digium TDM400P? We have a Digium TDM400P up and working with asterisk. We've fxotune'd the interface and pretty much eliminated all of the echo on the channel. Our latest issue is that all calls that run over the zap channels sound muffled and distant. I upped the rx gain and that helped some on the inbound voice but now we have some static creeping in. If I up the transmit gain, we start to get more echo so I'm really hesitant to do that too much. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] test
- Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 16, 2006 6:47 AM Subject: Re: [Asterisk-Users] Gumstix! James Harper wrote: http://www.gumstix.com For a non-telephony (Bluetooth based) project. I'm browsing the SVN website for Gumstix and lo and behold, there is Asterisk! I'm excited. Has anyone ever tried it on a GumStix before, and if so, care to share tips? I'd not heard of these before. Do you know if a BRI adapter can be obtained for them? Kristian at Astlinux is the person to talk to about these things. I think you can find out more about many aspects of embedded Asterisk at http://www.astlinux.org B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Large Asterisk System
Hey Guys I've been browsing the list looking for more information on asterisk behavior for large system. As for now I've got a project with 300 SIP Extensions to start, with future growth (scalability) Capability of recording all extensions simultaneously during peak time. And keeping the call recording for 30 days. SIP Calls being terminated over a Cisco 5400 Gateway Extra 100(+) FXO channels for incoming calls. (calls are going to be routed to the SIP Extensions. As far as I get the Hardware setup that I thought that will handle it is: Two Asterisk Server - Quad Dual-Core Opteron Servers running for call processing One SER Server - Dual Dual-Core Opteron for SIP routing and SIP 2 SIP calls (no recording) One Serial Over Ethernet Storage for recording the calls (the two asterisk servers will commit to that device) For the setup I was going to put a lot of RAM on the serves something like 8GB and make asterisk record the calls to a RAMDRIVE. Another process will run with low priority moving the Recordings from the RamDrive to the Storage. If Asterisk One dies, Asterisk two assumes. If SER dies, ASTERISK one or TWO will handle without the proxy. We need to avoid single point of failure as also be able to scale well. Other possibility that we may look is having instead of a storage, is having a Extra Asterisk getting the calls as a conference over IAX and saving it. As far as I see it will also need to deal with the real time recording on RAM. Because seek delay on multiple files being writing as small chunks of data (20ms voice data) on the HD will make voice choppy. So the solution will have to involve moving from RAM do HardDrive as the conversation ends. As I read it will be easy to record a 30megs file instead of several small chunks of 100KiloBits. As the other process can run in low priority and not realtime. It may not affect asterisk recording to RAM. Do you guys think that the servers will be able to handle that ? Does the SIP protocol can handle that redundancy ? Has anyone designed a system similar to this ? Does anyone wants to add a two cents comment on that design ? Is any company available for paid consulting ? -- Charles Rauber Gomes ___ ___ / //|/ / || / / // ) ) // ) ) // | | /__ ___/ / ///| / / || / / // / / ((//__| |/ / / /// | / / || / / // / /\\ / ___ | / / / /// | / /||/ / // / / ) ) //| | / / __/ /___ // |/ / | / ((___/ / ((___ / / // | | / / 954-585-1033 Extension 55 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Panasonic KXTD 1232 6
I want to replace a Telebutler software auto attendent system that used a 4 port Dialogic board connected to a Panasonic KXTD 1232 6 line system. We have spare computers here. How do I connect asterisk to this Panasonic system? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dumb question - reaching the PSTN
Hi everyone, I am fairly new to the idea of VoIP, although I've been reading about it off and on for the last few years. Now it is starting to look mature enough to consider implementing it, but there is one thing that I haven't been able to get a clear answer on... With Vonage, you are using the Vonage network - it is their responsibility to route your call to the endpoint, which is more than likely on the old fashined PSTN. If I install Asterisk, how do my calls actually get completed? How do they get 'bridged' over to the PSTN? I attended a Seminar today hosted by Dynasis, and one of the issues was VoIP. ShoreTel was there, and the said I had to have phone lines, whether they were POTS lines, chennels from a T-1, whatever, we still had to have phone lines. Now I'm confused. If I implement an Asterisk based system (yes, I'd be paying a consultant to help), will I still have to maintain phone lines and pay full price for Long Distance? Simple pointers to White Papers on this issue will be sufficient. Many thanks, -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Marketing Materials
Perhaps http://www.millenigence.com/articles/asterisk-non-technical-review.pdf ? Rich Adamson wrote: Darrick Hartman wrote: Bob McDowell wrote: The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but I think a pdf would be priceless at this point. Bob, Check on Digium's website. I know there is such a creature there. Darrick Just went looking and could not find a thing. Can you give us a url? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Recording?
Is Asterisk capable of allowing for the recording of calls on a per extension basis? -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Monitoring?
1. Is Asterisk capable of allowing for setting up Groups so that only one extension in a Group can selectively monitor one of the other extensions in the Group (but none of the others can initiate it)? This would be for Managers to listen to Sales Calls of other members of their Team, to provide feedback to the Rep for training purposes. 2. Alternatively, can a Group be defined that will allow multiple extensions to listen in on another call in progress? Again, we want to use this kind of functionality to do some Sales Technique Training calls. -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: problems with emailing voicemail
On 3/22/2006 Avi Miller ([EMAIL PROTECTED]) wrote: A smarthost is another SMTP server (e.g. your corporate email server, which should already be capable of sending outbound email) that your Asterisk box is configured to send all outgoing mail to, instead of trying to deliver it directly. Actually, that would more properly be called an SMTP RELAY (the SMTP server that Asterisk was talking to), would it not? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Recording?
Kris Edwards wrote: Absolutely. This is well documented on the wiki at voip-info.org http://voip-info.org (as are the possible legal issues that you may have by recording the calls). Search for info about the asterisk application Monitor. Great, thanks! We will be doing this with full knowledge of everyone participating in the call, so there will be no legal issues to worry about. Thanks again! -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Do you mean in general? Or only if you are trying to interconnect multiple offices? Are Polycoms fine for just one office, if the entire office is behind a NAT device, and the phones are only being used for normal calling? Thanks, -- Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
mustardman29 wrote: I might as well jump in. I am not clear on what the problem is but whether it's a problem on something that needs to be done frequently or infrequently or perhaps can be avoided with little effort, it's still a problem. Your argument is more like the classic it's not a bug, it's a feature. Well, I'll jump in too... If I had to 'reload' my postfix server, or my dovecot server, or my samba server all the time, I'd be wondering what the heck is the PROBLEM that is causing me to have to constantly reload (or restart) them. I agree that if there IS a problem with Asterisk that causes one to have to reload it often simply when making normal day-to-day changes (adding phones/users, etc), then I would certainly call that an Asterisk problem, but from what I've read, I don't think that is the case. Still, I'm new, and haven't even installed Asterisk yet - I'm not a 'phone' guy, and I've been driving my phone guy crazy trying to get him to start playing with it, so that we can start replacing our current system with it and some Polycom phones Charles. -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 22, 2006 9:46 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ? On Wednesday 22 March 2006 11:34, Douglas Garstang wrote: First thing that comes to mind, what if we decided to change a non user setting in sip.conf? You're reaching. You said you NEED to reload all the time, that this is a MAJOR issue, a deal breaker. So surely you must have experienced this downtime to be so sensitive to it. What did you do on your PRODUCTION system that required constant reloads to cause the current behavior to be such a big problem? Honestly; if you're changing a non-user setting in sip.conf you're going to do that very, very infrequently, and you'd do it during a low volume time. You said this is a major problem. I'm calling you on it. I'm interested in making Asterisk robust and highly-available too, but I'm not making up scenarios in order to launch complaints and verbal assaults against the project in order to feed my inflated ego and try to get things done my way. If you have a specific problem, let's hear it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
Whether or not a forum is a better idea isn't really depending on the subject matter IMHO. Its success or failure depends on what the prospective participants like better. I personally cannot stand forums. That's a place where I have to expend energy to go there and manually click through stuff. If I remember to go there and say up to date, that is. Email comes to me, and is sorted suitably on the server side so there is no clutter. Deleting messages I don't care about is much easier than clicking myself through some thread on a forum. You never heard of a forum that sends new posts to you via email? I prefer forums where I can subscribe to the forum topics that interest me, and see only posts for those topics - yes, in my email. Then each message from the forum should have links to the CENTRALIZED FAQs (I understand there are a lot of different forums/faq's out there). That said, I seem to be in the minority in preferring forums for supprt related things like this - especially high volume stuff - so I'll just pipe down now... :) -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] integration with Toshiba PBX system
Hi, I am currently integrating our company's Toshiba PBX with the Asterisk version 1.2.1. I bought Quad T1 card, and making the port 1 to connect to PSTN PRI (use pri_cpe in zaptel.conf) and making the port 3 to connect to Toshiba PBX (using pri_net in zaptel.conf). The first stage goal is to just adding the Asterisk relay between PSTN and Toshiba system. The issue I am facing is that I can make a outgoging call from Toshiba system phones to outside; but incoming calls always fail. I can observer the call come from span 1 and routes to span 3, but the call immediatedly hangup. Did anyone have experience on this issue. I try to make use the setting of misdn.conf to try to print out the signalling info, but it seems that there is no logging output. Is misdn.conf useful in 1.2.1 version. best regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
David Rahn wrote: This would lend its self to less repitition of questions as the lists would be much more searchable At this time I 3 months of this list and it is over 13,900 messages. In other words GREAT IDEA I THIRD THAT!! I do think all hardware disscussion ( as it effects Asterisk) should be grouped togeather. As it is not always the exact same problem that is what helps to fix your problem ... Actually, for something like Asterisk, that has so many different aspects, a Forum would be a much better idea. Then, each piece of hardware can have its own category, along with an FAQ. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New ncurses Asterisk Manager Interface
On 3/16/2006 El Flynn ([EMAIL PROTECTED]) wrote: I'm trying to compile the assman package This is a jok, right? I mean, no one would actually name a project something like 'assman', would they? lol ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: regexten
On 3/17/2006 Douglas Garstang ([EMAIL PROTECTED]) wrote: It doesn't matter how often I expect them to go down. You have to plan for the worst. People are used to say, being unable to access a web page for a few minutes. Heck, they might not even notice. However, when someone's phone system goes down, you can bet your bottom dollar that they will notice pretty damn fast, and they expect it to always be working. There isn't a phnoe/phone system in the WORLD that 'always works'. 100% uptime is IMPOSSIBLE. The cost increases exponentially, the closer to 100% you want to get, but you will never get there. I think 5 9's is perfectly acceptable, even for the most demanding, high-powered executive. -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toshiba Strata DK-280 support?
Philip Edelbrock wrote: At some point (in a few months, probably) we'll turn off the Toshiba and put viop phones on everyone's desk (including some people's at a remote office and homes). It should also cut our phone bill down to a 1/10th of what it is now! Interesting... so, you consider Asterisk / VoIP secure enough at its current stage? I have heard a lot of horror stories, and as well, I have actually experienced firsthand how bad the quality can be (I have Vonage at home, and I have had conversations from our phone system in our office with people who had VoIP systems, and the quality was pretty bad (sounded like they were underwater). This is definitely something that interests me, but I'd also be very interested in hearing others experiences with VoIP - anyone? -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Toshiba Strata DK-280 support?
Hi everyone, Been reading up on Asterisk, and very interested in learning more. I've googled and read the archives and haven't found anything definitive on support for this phone system. We have a fairly large investment in the system itself and the phones, but would love to get away from the voicemail system it forces on us. Can anyone provide any feedback on using this system with Asterisk? Am I wasting my time even thinking about it? Thanks, -- Best regards, Charles Marcus I.T. Director Media Brokers International 678.578.2200 x224 678.578.2240 fax ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toshiba Strata DK-280 support?
I run Asterisk partnered up to the 280 (424 for us). We have a 6 cabinet installation of the Toshiba so I understand you dilemma. There are some quirks with the 280 that make it a challenge to use Asterisk with, but it's do-able. Keep in mind, you can move to the CTX or the CIX and still keep a lot of your investment and those systems play much better with Asterisk. Hi Brian, Thanks for your response. I am investigating Asterisk as a possibility for our company, but I am not a 'phone guy', I am the Network Admin. I sent our phone guy a link to the Asterisk site, but he hasn't had time to look at it yet. I am simply trying to get a head start for him. He seemed to think it highly unlikely that we would be able to use any of our current system with it, but he admitted that not knowing anything about Asterisk, it was certainly possible he was wrong. So, that said - after reading a little on CTX/CIX - those are just VoIP, but is this just a card added to our cabinet, then we need to get all new phones? We are not that big of a company. We only have 35 phones now, with a potential to double or so in the next year or two. I really don't see us growing past 100 phones in the next 5 years, and our plan is to have the company sold well before then. We don't have a huge need for VoIP, although the boss may decide to give that a try some day. I still have concerns about reliability/voice quality/security myself. So, how much work are we talking about to get our current system to play nice with Asterisk? Will we lose any functionality? Gain any? Do you know of any technical how-to's that my phone guy would be able to answer these questions from? Are you available to concult? If so, for how much? Sorry to hit you with so much, but if I don't ask... ;) -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7970 Configs
To get the the dialplan working change: dialTemplate/dialTemplateto: dialTemplatedialplan.xml/dialTemplateand place a dialplan.xml file in your tftp directory.Simple dialplan.xml file:DIALTEMPLATE TEMPLATE MATCH="*" Timeout="15"//DIALTEMPLATEalso to activate the 7914 add:addOnModulesaddOnModule idx="1"loadInformationS00105000100/loadInformation/addOnModule!-- Uncomment if you have second 7914addOnModule idx="2"loadInformationS00105000100/loadInformation/addOnModule--/addOnModulesjust after the loadInformation tag. You will need to load the S00105000100 file toyour tftp directory.The things I cannot figure out are: 1. How to set the secret for proxy registration. 2. How to define speeddials.Thanks,/canIf I recall when we first got the CCM5 development SIP loads, I got thesame result, but it was funny that * showed the phone as not registered.It may well be the fact that I have not downloaded the released version.It may be more non-CCM friendly.I'll play with it again next week if I can borrow a 70 away from thedevelopers for a while.The only thing I do not like about the 41/61/70/71 (all the java phones)is they only allow one password for all the separate lines/proxies inSIP mode. I may play with the config to see if it will allow more.-GregBTW: If you do get it to play nice, please post the xml file for us :)On Fri, 2006-03-10 at 13:56 -0600, Aaron Daniel wrote: Awesome, that works, 'cept now the dialplan doesn't work lol. I've programmed the voicemail button in, but anything I try to dial doesn't make it past the first digit. Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers
Hi, ALL: Can anyone tell me what *RT is ? What is its full name? I think the * is asterisk but what is RT ? 2006/2/2, Rusty Shackleford [EMAIL PROTECTED]: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: Wednesday, January 04, 2006 4:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers load balacing isn't perfect, and it can give uneven loads at low capacity, but it gets better as load increases which is where it matters. What kind of loads are we talking about here, please? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Serusers] high-availibility setup using f5 bigip
I think that the range of this question is too large. You should tell us what your scenario is. And tell us more about your configurations. 2006/2/2, Jack Wei [EMAIL PROTECTED]: hi, I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do load-balancing. I'm using Asterisk as a voicemail application only and have successfully integrated SER with Asterisk without the switch. But when I try to use the switch as a load-balancer, I get lots of NAT problems. Does anyone know how to setup the switch and SER/Asterisk properly? Thanks, Jack __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers -- Best Regards Charles -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the none commercial intel g729codecs into [EMAIL PROTECTED] 2.2?
I have the same problem too. I install the G.729 (IPP) to asterisk 1.0.x, and it works well. When I change asterisk from 1.0.x to 1.2.x, and G.729 seems work fine. I can use show translation and find it too. But when I make a call using G.729. The asterisk (1.2.1) crashed. If i mark the line allow=g729 from sip.conf. And asterisk works fine. 2006/1/22, Guillermo Salas M [EMAIL PROTECTED]: Con fecha 21/1/2006, Francesco Peeters (Asterisk) [EMAIL PROTECTED] escribió: On Sat, January 21, 2006 23:21, Franz Bräuer said: Hi, MapsAir wrote: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? I'm using g723.1 and works very well. Installed them today. Installing from source didn't work for me (Debian, Asterisk 1.2 from svn) but just adding the binaries (see the wiki on voip.org) did the job. Have you already tried the binaries? Kewl! Those work like a treat! As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did: cd /usr/lib/asterisk/modules/ wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so After reloading, 'show translation' gives: Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 -22 8 817 8 724 115 19897 gsm 151 - 7 716 7 623 114 19796 ulaw 14616 - 111 2 118 109 19291 alaw 14616 1 -11 2 118 109 19291 g726 154241010 -10 926 117 20099 adpcm 14616 2 211 - 118 109 19291 slin 14515 1 110 1 -17 108 19190 lpc10 161311717261716 - 124 207 106 g729 16939252534252441 - 215 114 speex 16030161625161532 123 - 105 ilbc 17343292938292845 136 219 - Jolly good show, old chap! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Suggestions for E1 Service in the UK
snip Well, the major incumbent is BT. Are you sitting down ? Installation : Per channel 1 year contract 3/5y contract 3/5y+commitment First 15 channels (min 8)GBP 125 GBP 80GBP 0 16-30 (per channel) GBP 30 GBP 15GBP 0 Annual Rental (per channel) GBP 182.32 DDI Non Quota GBP 208.32 DDI Quota jd Affiniti (Kingston Communication) is another choice. Their min channel number is 6, and their pricing is more reasonable than BT. Dont have exact prices to hand, but they are better. They are also a far easier company to deal with than BT, who I have had no end of problems with. Charlie ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recommendations for * monitoring?
Hello, Can anyone point me in the direction of software to monitor channel usage on voice T1s? Using a TE410. The wiki documentation seems geared to SIP channel usage Thanks Charles ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GalaxyVoice Problems
Has anyone had a problem with GalaxyVoice in making international calls from the USA? Every single call that I attempt to Ireland, England, Japan, etc is met with either an instant congestion/all circuits busy OR it rings about seven times and then drops off saying that Everyone is busy/congested at this time. I try these calls at all times of the day and evening with the same results. I have my * set to allow ALL codecs. -- Charles J. Hargrove - N2NOV NYC ARECS/RACES Citywide Radio Officer/Skywarn Coord. US Coast Guard Auxiliary Flotilla 5-10 Comms Officer NYC-ARECS/RACES Net Mon. @ 8:30PM 147.360/107.2 PL http://www.nyc-arecs.org and http://www.nyc-races.org NYDXA SWL Scanner Net Wed. @ 9PM 147.360/107.2 PL http://www.n2nov.net Information is the oxygen of the modern age. It seeps through the walls topped by barbed wire, it wafts across the electrified borders. - Ronald Reagan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Malformed/Missing.URL Error from CallManager
Hi, I setup a SIP trunk between asterisk and Cisco CallManager according the wiki page. http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration But I'm getting a 'Malformed/Missing URL' from the CallManager. Does anyone know what went wrong here? I'm running asterisk CVS HEAD and (192.168.1.5 five) Cisco Callmanager 4.0(2a) (192.168.1.101) below is the debug from asterisk. Thanks for all your help. Regards, Charles five*CLI 11 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.1.101:5060: OPTIONS sip:192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK03c2094d From: asterisk sip:[EMAIL PROTECTED];tag=as4f7fa56a To: sip:192.168.1.101 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 23 Jun 2005 05:07:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 --- five*CLI -- SIP read from 192.168.1.101:5060: SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK03c2094d From: asterisk sip:[EMAIL PROTECTED];tag=as4f7fa56a To: sip:192.168.1.101 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Content-Length: 0 Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error on installing oh323 on asterisk
I'm following the instruction from João Amaro from the page http://lists.digium.com/pipermail/asterisk-users/2005-February/090752.html Everything went fine until I run the 'make' command under asterisk-oh323-0.6.5. I got the error message chan_oh323.c:5220: too many arguments to function `ast_channel_register' I have attached the error message. I'm running asterisk CVS HEAD version, would that be the cause of the problem? Any help would greatly appricated. Thanks, Charles # make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/root/asterisk-oh323-0.6.5/wrapper' ./check_ver /root/pwlib pwlib ./check_ver /root/openh323 openh323 ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/root/asterisk-oh323-0.6.5/wrapper' make[1]: Entering directory `/root/asterisk-oh323-0.6.5/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_exception': chan_oh323.c:1145: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_indicate': chan_oh323.c:1326: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_digit': chan_oh323.c:1388: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_text': chan_oh323.c:1410: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_call': chan_oh323.c:1434: dereferencing pointer to incomplete type chan_oh323.c:1453: structure has no member named `callerid' chan_oh323.c:1455: structure has no member named `callerid' chan_oh323.c:1457: structure has no member named `callerid' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1613: dereferencing pointer to incomplete type chan_oh323.c:1721: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_read': chan_oh323.c:1749: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_write': chan_oh323.c:2050: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_answer': chan_oh323.c:2242: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_fixup': chan_oh323.c:2286: dereferencing pointer to incomplete type chan_oh323.c: In function `ast_oh323_new': chan_oh323.c:2518: dereferencing pointer to incomplete type chan_oh323.c:2527: dereferencing pointer to incomplete type chan_oh323.c:2529: dereferencing pointer to incomplete type chan_oh323.c:2536: dereferencing pointer to incomplete type chan_oh323.c:2537: dereferencing pointer to incomplete type chan_oh323.c:2538: dereferencing pointer to incomplete type chan_oh323.c:2539: dereferencing pointer to incomplete type chan_oh323.c:2540: dereferencing pointer to incomplete type chan_oh323.c:2541: dereferencing pointer to incomplete type chan_oh323.c:2542: dereferencing pointer to incomplete type chan_oh323.c:2543: dereferencing pointer to incomplete type chan_oh323.c:2544: dereferencing pointer to incomplete type chan_oh323.c:2545: dereferencing pointer to incomplete type chan_oh323.c:2546: dereferencing pointer to incomplete type chan_oh323.c:2547: dereferencing pointer to incomplete type chan_oh323.c:2548: dereferencing pointer to incomplete type chan_oh323.c:2549: dereferencing pointer to incomplete type chan_oh323.c:2550: dereferencing pointer to incomplete type chan_oh323.c:2551: dereferencing pointer to incomplete type chan_oh323.c:2552: dereferencing pointer to incomplete type chan_oh323.c:2579: structure has no member named `dnid' chan_oh323.c:2589: structure has no member named `callerid' chan_oh323.c:2590: structure has no member named `callerid' chan_oh323.c:2591: structure has no member named `callerid' chan_oh323.c:2596: structure has no member named `callerid' chan_oh323.c:2597: structure has no member named `callerid' chan_oh323.c:2598: structure has no member named `callerid' chan_oh323.c:2600: structure has no member named `callerid' chan_oh323.c:2605: structure has no member named `callerid' chan_oh323.c:2606: structure has no member named `callerid' chan_oh323.c:2608: structure has no member named `callerid' chan_oh323.c:2610: structure has no member named `callerid' chan_oh323.c:2614: structure has no member named `callerid' chan_oh323.c:2617: structure has no member named `ani' chan_oh323.c:2617: structure has no member named `callerid' chan_oh323.c:2623: structure has no member named `callerid' chan_oh323.c:2624: structure has no member named `callerid' chan_oh323.c: In function `oh323_request': chan_oh323.c:2741: dereferencing pointer to incomplete type chan_oh323.c:2743: dereferencing pointer to incomplete type chan_oh323.c: In function `oh323_atexit': chan_oh323.c:4923: warning: passing arg 1 of `ast_channel_unregister
Re: [Asterisk-Users] SIP_HEADER - anybody using it?
Where is the function? On source codes or any config file? On 6/14/05, Denis Galvão - iSolve [EMAIL PROTECTED] wrote: Hi all. Could someone point me an example to use SIP_HEADER function!? I want to read the To: and send this INVITE to an internal extension. Is there anybody using this function!? Tks. Denis Galvão ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Please comment on Dvorak's troll
On 6/7/05, Michael Graves [EMAIL PROTECTED] wrote: On Mon, 6 Jun 2005 11:17:20 -0600, Colin Anderson wrote: http://www.pcmag.com/article2/0,1759,1812887,00.asp Specifically, his assertion that ISP's would sniff traffic and block, say, the SIP port. You could play wack-a-mole with port numbers, no? Also a community based, Freenet style of encryption implementation for free VoIP traffic would address this issue. I raise this to the list because I'm sure there's a grain of truth in what he's saying. ILEC's would be crazy to not consider this kind of lock in, since it's pretty obvious that packet voice networks are going to supplant circuit networks completely in, say, 20 years. Maybe sooner. Actually, Bob Cringley, another pundit found on the PBS web site raised this matter a few weeks ago. I suspect that IAX2 with some encryption could port hop around and not be easily tracked as VOIP traffic. But in any case there has to be some regulatory stance on what is permitted over a network. Certainly there are non-telco carriers like Covad, whom I use, that would not concern themselves about the nature of the traffic. Michael -- Actually, the FCC has already come down hard on an independent phone company that blocked VoIP traffic for a number of years. Vonage complained, and finally won: http://www.pcpro.co.uk/news/70081/us-slaps-fine-on-company-blocking-voip.html The telcos have seen this coming for years, and many of them are getting into the Video over DSL space as a means to compete going forward. Off topic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie question
Greetings, I have my first asterisk installation up and running, thanks to a lot of reading. Could anyone point me in the direction of things to read on automated outbound dialing? NOT predictive dialing - I will not have agents handling the calls. These calls are reminders for appointments, etc. Thanks! Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Areskicc v2 login issues
Hello, I have modified php.ini to have global variables but it still does not log in.. I have also installed php-pgsql... so I should be fine.. Anyways I cannot login "Invalid login/password...". Any other hints? Is support available.. I really want toinstall this application. Thanks in advance, regards, ACYahoo! Mail - Votre e-mail personnel et gratuit qui vous suit partout ! Créez votre Yahoo! Mail sur http://mail.yahoo.fr___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Invalid login/password with AreskiCC V2
Hi Everybody, I have tried to make AreskiCCV2 work on RH9.0 but it does not work. More precisely, I have followed the guide as well as the installation instructions but I always getan Invalid login/password error when i try to login using the web interface. The login/password provided do match in all the configuration files. Any clues? Any comments on the applications? Any alternative to the application? Thanks in advance, ACLèche-vitrine ou lèche-écran ? Yahoo! Magasinage.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP PLZ$B!'(Bsip channel AGI problem
Hi, ALL:
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And this 0939749001 will be sent to asterisk by sip proxy.
sip ua(1011) = sipproxy = sip ua ( call forward 0939749001)
||
== asterisk === cisco 5300 ==
0939749001 (pstn)
I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of from To: field? ( via this sip ua)
In another word, I want to record the middle man.
My extensions.conf :
exten = _.,1,Answer
exten = _.,2,DeadAGI(my.agi,${CALLERIDNUM},${EXTEN})
exten = _.,3,Hangup
My log on asterisk CLI:
-- Executing DeadAGI(SIP/61.220.xxx.xxx-081888c8,
my.agi|1011|0939749001|4) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI
-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: sip:61.220.xxx.xxx;ftag=915860198;lr=on
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 sip:[EMAIL PROTECTED];tag=915860198
To: sip:[EMAIL PROTECTED];tag=as1c0a7e38=== I want to get this value
Contact: sip:[EMAIL PROTECTED]:47286
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0
--
Best Regards
Charles
--
Best Regards
Charles
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: HELP: ASTCC (AGI) meets call forward ERROR
On 5/12/05, Charles Wang [EMAIL PROTECTED] wrote:
Hi, ALL:
When I use astcc to do the prepaid function, but if I want to enable
call forward.
The result of CDR seems not correct.
UA 1011 make a call to UA , and UA forwards this call to a PSTN
number.
I think we shall charge the credit from UA not UA 1011 because UA
1011 don't know where UA forwards to.
But in CDR, I can only find the from(1011) and destination(PSTN number).
I can't find UA from this CDR so I can't charge to UA .
It seems unreasonable.
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And this 0939749001 will be sent to asterisk by sip proxy.
sip ua(1011) = sipproxy = sip ua ( call forward 0939749001)
||
== asterisk === cisco 5300 ==
0939749001 (pstn)
I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of from To: field? ( via this sip ua)
In another word, I want to record the middle man.
My extensions.conf :
exten = _.,1,Answer
exten = _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten = _.,3,Hangup
My log on asterisk CLI:
-- Executing DeadAGI(SIP/61.220.xxx.xxx-081888c8,
astcc.agi|1011|0939749001|4) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI
-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: sip:61.220.xxx.xxx;ftag=915860198;lr=on
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 sip:[EMAIL PROTECTED];tag=915860198
To: sip:[EMAIL PROTECTED];tag=as1c0a7e38=== I want to get this value
Contact: sip:[EMAIL PROTECTED]:47286
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0
--
Best Regards
Charles
--
Best Regards
Charles
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] $B#H#E#L#P!'(Bsip channel AGI problem
Hi, ALL:
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And this 0939749001 will be sent to asterisk by sip proxy.
sip ua(1011) = sipproxy = sip ua ( call forward 0939749001)
||
== asterisk === cisco 5300 ==
0939749001 (pstn)
I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of from To: field? ( via this sip ua)
In another word, I want to record the middle man.
My extensions.conf :
exten = _.,1,Answer
exten = _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten = _.,3,Hangup
My log on asterisk CLI:
-- Executing DeadAGI(SIP/61.220.xxx.xxx-081888c8,
astcc.agi|1011|0939749001|4) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI
-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: sip:61.220.xxx.xxx;ftag=915860198;lr=on
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 sip:[EMAIL PROTECTED];tag=915860198
To: sip:[EMAIL PROTECTED];tag=as1c0a7e38=== I want to get this value
Contact: sip:[EMAIL PROTECTED]:47286
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0
--
Best Regards
Charles
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: ASTCC (AGI) meets call forward ERROR
Hi, ALL:
When I use astcc to do the prepaid function, but if I want to enable
call forward.
The result of CDR seems not correct.
UA 1011 make a call to UA , and UA forwards this call to a PSTN number.
I think we shall charge the credit from UA not UA 1011 because UA
1011 don't know where UA forwards to.
But in CDR, I can only find the from(1011) and destination(PSTN number).
I can't find UA from this CDR so I can't charge to UA .
It seems unreasonable.
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And this 0939749001 will be sent to asterisk by sip proxy.
sip ua(1011) = sipproxy = sip ua ( call forward 0939749001)
||
== asterisk === cisco 5300 ==
0939749001 (pstn)
I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of from To: field? ( via this sip ua)
In another word, I want to record the middle man.
My extensions.conf :
exten = _.,1,Answer
exten = _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten = _.,3,Hangup
My log on asterisk CLI:
-- Executing DeadAGI(SIP/61.220.xxx.xxx-081888c8,
astcc.agi|1011|0939749001|4) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI
-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: sip:61.220.xxx.xxx;ftag=915860198;lr=on
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 sip:[EMAIL PROTECTED];tag=915860198
To: sip:[EMAIL PROTECTED];tag=as1c0a7e38=== I want to get this value
Contact: sip:[EMAIL PROTECTED]:47286
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0
--
Best Regards
Charles
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: HELP: how to get To: from AGI?
On 5/9/05, Charles Wang [EMAIL PROTECTED] wrote:
Hi, ALL:
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And this 0939749001 will be sent to asterisk by sip proxy.
sip ua(1011) = sipproxy = sip ua ( call forward 0939749001)
||
== asterisk === cisco 5300 ==
0939749001 (pstn)
I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of from To: field? ( via this sip ua)
In another word, I want to record the middle man.
My extensions.conf :
exten = _.,1,Answer
exten = _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten = _.,3,Hangup
My log on asterisk CLI:
-- Executing DeadAGI(SIP/61.220.xxx.xxx-081888c8,
astcc.agi|1011|0939749001|4) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI
-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: sip:61.220.xxx.xxx;ftag=915860198;lr=on
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 sip:[EMAIL PROTECTED];tag=915860198
To: sip:[EMAIL PROTECTED];tag=as1c0a7e38=== I want to get this value
Contact: sip:[EMAIL PROTECTED]:47286
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0
--
Best Regards
Charles
--
Best Regards
Charles
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BYE from Cisco gateway
And I find that my cisco will send BYE after 30 seconds after PSTN hangup. On 5/11/05, Charles Wang [EMAIL PROTECTED] wrote: yes, my cisco trunking gateway has also this problem. On 5/11/05, Torbjørn Lium [EMAIL PROTECTED] wrote: What makes you think I'm not trying a cisco user list? At least it's worth a try to post the question here also. C F wrote: Why don't you try a cisco user list? On 5/10/05, Torbjørn Lium [EMAIL PROTECTED] wrote: I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN. If a user on a softphone hangs up first the PSTN port on the cisco is released and new calls can be made on the same voice port. But when the user on the PSTN side hangs up first the voice port on the cisco stays open until the user on the softphone hangs up. Any ideas what I'm doing wrong? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BYE from Cisco gateway
yes, my cisco trunking gateway has also this problem. On 5/11/05, Torbjørn Lium [EMAIL PROTECTED] wrote: What makes you think I'm not trying a cisco user list? At least it's worth a try to post the question here also. C F wrote: Why don't you try a cisco user list? On 5/10/05, Torbjørn Lium [EMAIL PROTECTED] wrote: I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN. If a user on a softphone hangs up first the PSTN port on the cisco is released and new calls can be made on the same voice port. But when the user on the PSTN side hangs up first the voice port on the cisco stays open until the user on the softphone hangs up. Any ideas what I'm doing wrong? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: HELP: how to get To: from AGI?
On 5/9/05, Charles Wang [EMAIL PROTECTED] wrote:
Hi, ALL:
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And this 0939749001 will be sent to asterisk by sip proxy.
sip ua(1011) = sipproxy = sip ua ( call forward 0939749001)
||
== asterisk === cisco 5300 ==
0939749001 (pstn)
I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of from To: field? ( via this sip ua)
In another word, I want to record the middle man.
My extensions.conf :
exten = _.,1,Answer
exten = _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten = _.,3,Hangup
My log on asterisk CLI:
-- Executing DeadAGI(SIP/61.220.xxx.xxx-081888c8,
astcc.agi|1011|0939749001|4) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI
-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: sip:61.220.xxx.xxx;ftag=915860198;lr=on
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 sip:[EMAIL PROTECTED];tag=915860198
To: sip:[EMAIL PROTECTED];tag=as1c0a7e38=== I want to get this value
Contact: sip:[EMAIL PROTECTED]:47286
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0
--
Best Regards
Charles
--
Best Regards
Charles
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] detaching console from background asterisk
This puzzles me. If I start asterisk in the background, and then attach to it to perform some chores, is there a way to detach again without stopping the background process? Entering stop now kills both the console attachment as well as the background process. I need to attach to the running asterisk in order to do init keys but once I do that, it seems I cannot just let it go into the background again. Any suggestions most welcome. Chuck -- The Moon is New Malt does more than Milton can, To justify God's ways to Man. You can download some things from http://www.mhcable.com/~chuckh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] detaching console from background asterisk
Super! Just what I needed. Many thanks. On Sun, 8 May 2005, Ron Wellsted wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Charles Hallenbeck wrote: This puzzles me. If I start asterisk in the background, and then attach to it to perform some chores, is there a way to detach again without stopping the background process? Entering stop now kills both the console attachment as well as the background process. I need to attach to the running asterisk in order to do init keys but once I do that, it seems I cannot just let it go into the background again. Any suggestions most welcome. try quit - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD:519961 Gossiptel:9309811 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iQEVAwUBQn5sd0tP/KMNOfRbAQIyxgf+KaIOqmYWihrOqwmPKtrdKsFvq5t7LAw0 jLQZ8TcY3sNQ64tLT8nw3ZTlLUielzeJ69rO7gQWBH5VxqZc6az22nSbr0dmHHlI exgqODttcUyqK635OUIImFXFmLSdxGKFo7bZ+9tXygyQSjJexU5mwqcrfBft0kYk 9CqCAF2/AFxUgNVcWl4qOCIDmGIZavMx4nl8GTPdZTuSZV4UTH1P7WhCPMGoPRF5 DxxB84UEMkm2Hp1DkwYXBfctySqOgponFPfpqHdsN67z7AA4XCEuQKf9kfdIZ0QD CaQUGxk1+O2nTcT+fqRc53JHJclMVi4A6wkrIzOPNWXMjVUHVBTXjQ== =MFqH -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- The Moon is New Malt does more than Milton can, To justify God's ways to Man. You can download some things from http://www.mhcable.com/~chuckh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: how to get To: from AGI?
Hi, ALL:
I use asterisk -r and sip debug to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to 0939749001.
And this 0939749001 will be sent to asterisk by sip proxy.
sip ua(1011) = sipproxy = sip ua ( call forward 0939749001)
||
== asterisk === cisco 5300 ==
0939749001 (pstn)
I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of from To: field? ( via this sip ua)
In another word, I want to record the middle man.
My extensions.conf :
exten = _.,1,Answer
exten = _.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
exten = _.,3,Hangup
My log on asterisk CLI:
-- Executing DeadAGI(SIP/61.220.xxx.xxx-081888c8,
astcc.agi|1011|0939749001|4) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI
-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route: sip:61.220.xxx.xxx;ftag=915860198;lr=on
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 sip:[EMAIL PROTECTED];tag=915860198
To: sip:[EMAIL PROTECTED];tag=as1c0a7e38=== I want to get this value
Contact: sip:[EMAIL PROTECTED]:47286
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0
--
Best Regards
Charles
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] email notification when leaving a message
Hi! I have configured: iax.conf; voicemail.conf extensions.conf everything works fine... the only things.. i do not receive any email notification when a voicemail is left on the *.. any clues??? i think my email server works(?).. In fact i am able to send an email to the root (mail root etc...).. but aside that.. i am not able to send any other email outside the * box... any clues on how to solve that... I have installed * on RedHat.. Thanks for your help, AlexC __ Lèche-vitrine ou lèche-écran ? magasinage.yahoo.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registerport 5060 or 1720?
The 5060 is usually SIP Proxy listen port. And the 1720 is usually h323 gatekeeper's listen port. On 4/24/05, Ronald Wiplinger [EMAIL PROTECTED] wrote: When do you use Registerport 5060 and when 1720 ?? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: HELP: How to detect a hangup tone?
On 4/19/05, Charles Wang [EMAIL PROTECTED] wrote: Dear ALL: My scenario is: SIP UA == SIP Proxy == Asterisk == CISCO 5300 trunk == PSTN I make a call from SIP UA to a PSTN phone number, and SIP UA hangup first. My Asterisk can receieve a BYE message, so this connection will be hangup. But if my PSTN side hangup first, my CISCO will send BYE to Asterisk after 30 seconds. And Asterisk disconnects this connection at this time(receives a BYE via CISCO). Does anyone have solution/idea to make asterisk hangup immediately? How to change the configuration of CISCO and send a BYE immediately or Asterisk can detect a hangup tone? -- Best Regards Charles -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: How to detect a hangup tone?
Dear ALL: My scenario is: SIP UA == SIP Proxy == Asterisk == CISCO 5300 trunk == PSTN I make a call from SIP UA to a PSTN phone number, and SIP UA hangup first. My Asterisk can receieve a BYE message, so this connection will be hangup. But if my PSTN side hangup first, my CISCO will send BYE to Asterisk after 30 seconds. And Asterisk disconnects this connection at this time(receives a BYE via CISCO). Does anyone have solution/idea to make asterisk hangup immediately? How to change the configuration of CISCO and send a BYE immediately or Asterisk can detect a hangup tone? -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMP for over 1000 dollars
Is this ok to sell this on Ebay when they are using open source software? http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItemitem=5766004579 Hoping to have helped, Charles Osstyn 11, Cowper Crescent Foxhill, Sheffield S6 1AU United Kingdom Standard contact channels Tel +44 (0)114 231 38 98 (Now connected to our VOIP server Gonzo, hit option five for my office line.) Mob +44 (0)790 393 91 46 Fax +44 (0)870 051 79 92 Preferred VOIP contact channels SIP sip:[EMAIL PROTECTED] (Get a free pre-configured (with account) VOIP soft phone from FWD (Free World Dialup) here for your PC, laptop or Pocket PC.) E-mail [EMAIL PROTECTED] Webwww.osstyn.com Webcam www.osstyn.com:81/guest.htm On request via Skype. Skype charelke (Get the free Skype VOIP client here.) MSN Messenger [EMAIL PROTECTED] (Get MSN Messenger here.) E-MAIL DISCLAIMER The information in this e-mail is confidential, and is intended solely for the addressee's. Access to this email by anyone else is unauthorised and therefore prohibited. If you are not the intended recipient, or if the email is marked as 'confidential', any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration to multiple GKs
Is it possible to run Asterisk with another GKs using Neighbor mode? If it is possible, we can run asterisk with several gnugks. On Apr 2, 2005 10:41 PM, Alex Vishnev [EMAIL PROTECTED] wrote: I don't think you can. The rules of h323 is so that you can register with a single gk at a time. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie Sent: Saturday, April 02, 2005 6:37 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Registration to multiple GKs Hi all, How can I configure chan_h323 or chan_oh323 to register to multiple GK and route calls in-between? Many thanks. Newbie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] V92 modem with asterisk
Hi everyone, I just install Linux and asterisk on one of my pc. I want to run some basic functionality tests. Is it possible to use a v92 modem as a FXO or FXS card. If yes how do we configure and install the card? What are the commands? Thanks in advance for your help AC __ Lèche-vitrine ou lèche-écran ? magasinage.yahoo.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: How to configure h323 channel driver ?
Hi, ALL:
I has installed my chan_h323 channel driver in my *.
my scenario is:
SIP UA = SER(mediaproxy) = Asterisk = chan_h323 = GNUGK = H323 EP
And my UA and EP all support codecs such as alaw ulaw G.729 at least.
I dial from UA behind NAT to H323 EP, and I answer from H323 EP too.
But I can not hear any voice from each side. Can anybody point out why it is?
h323.conf
--
[general]
port = 1720
bindaddr = 0.0.0.0
tos=lowdelay
accountcode = myaccountname
gatekeeper = IP of GNUGK
AllowGKRouted = yes
amaflags=default
type=h323
prefix=888248
e164=8881238
context=voip323
disallow=all
allow=g729
allow=gsm
allow=alaw
allow=ulaw
allow=g723.1
extensions.conf
--
[general]
static=yes
writeprotect=no
[globals]
[default]
exten = _.,1,Dial(H323/${EXTEN})
--
Best Regards
Charles
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable
Hi, ALL: I'm almost give up the oh323 too. I compiled the asterisk-oh323 for several times(ten or more). In my first time, I got pwlib 1.6.6 from CVS of openh323.org. But it seems a little buggy,so I failed. I downloaded Janus's patch version, and followed its steps. It seems OK when I compile with pwlib 1.6.6 and openh323 1.13.5 plus oh323 v0.7.1. But I want my Asterisk to transfer my SIP call to H323 gatekeeper. SIP UAC == SER == Asterisk == GNUGK =X= some H323 EPs. There are too few documents or mailling lists to tell me how to configure it. I want my asterisk registering to GNUGK as a gateway mode. I know how to set it up on my GNUGK's gatekeeper.ini. But does anyone kind to tell me how to configure my extensions.conf and oh323.conf? I can make a call to reach GNUGK. But the call will be hangup for some reasons(I don't know what reason it is) when I find ACF and unconnected CDR on my GNUGK's log. There is NOT any EP rings at that time. It is very difficult to setup such a environment and too few users discuss about it. Is the way just give it up? On Wed, 23 Mar 2005 19:33:55 +0100, Yves [EMAIL PROTECTED] wrote: Try to isolate the problems, and send bugs to : https://skylab.inaccessnetworks.com/mantis/main_page.php Doing this will improve the project. We're using it and it's working pretty good. Don't give up too fast! Yves Bashir Ullah - www.Lamsre.Com wrote: Hi George I did install and checkup several times, but some times h323 gateway or softswitch cant accept my call and was able to accept call but no sound. so can you help me please to implement a h323 solution. You may contact with me if you want. Thanks Bashir Call. 1-604 323 7991 Mail. [EMAIL PROTECTED] - Original Message - From: George K. Konstantoulakis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 3:11 AM Subject: Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable Hello Bashir, what kind of problems are you having with oh323 ? George Bashir Ullah - www.Lamsre.Com wrote: Hi All * lover. This is not a question only this is a request to all SIP and Asterisk user . I am also with asterisk last few month and providing callingcard soluation. most of the SIP or IAX provider asking very high price which is really tough to resell in market. but still there is some h323 provider offering good price. so as a asterisk user i tried so many times and now give up to implement oh323, h323 by asterisk. i am sorry and also there is very may be none user for asterisk with h323. Thats why need a seperate soluation and open source for converter h323 to sip vies-versa for asterisk user. Is it possible in near future. or is there any solution already done with is open source. Thanks for your time to read this mail. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
