Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4
Hi Atis, queue_log = mysql,asteriskcdrdb,queue_log that is engine,database,table If it's wrong, you should see some warnings when asterisk is starting up. Thanks for the suggestion. I did not put in queue_log for table and it has just taken the default which is queue_log. In the console startup, you can see below that it has successfully bound queue_log to /mysql/db1/queue_log. # asterisk -rvvv Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. [...] == Parsing '/etc/asterisk/extconfig.conf': Found == Binding queue_log to mysql/db1/queue_log Connected to Asterisk 1.4.21.2 currently running on machine Verbosity is at least 3 In /var/log/asterisk/messages, I saw: [Oct 15 15:31:48] NOTICE[20941] config.c: Registered Config Engine mysql Another idea that came into my mind is, that (if this config doesn't still work) you might have to do make dist-clean within asterisk-addons after reinstalling asterisk, and then configure, make, make install. It's because addons do use headers from installed version of asterisk, and they might not have correct declarations. Basically, I did: - Asterisk-1.4.21.2 make clean ./configure make make install - Asterisk-addons-1.4.7 make dist-clean ./configure make make install Also, you mentioned that you checked /var/log/asterisk/messages, however i think debug is written into file called debug. Anyway you can enable full in logger.conf and get everything there. To debug this you shouldn't need more than core set verbose 3 and core set debug 1. I turned on debug mode and tried an agent login and logoff. However, when I looked into debug and messages, there are lots of chan_sip.c and a few cdr_addon_mysql.c but no occurrence at all of res_config_mysql.c What is happening? Do I have to explicitly load it? *CLI module show like res_config_mysql Module Description Use Count res_config_mysql.so MySQL RealTime Configuration Driver 0 1 modules loaded ___ -- 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] FW: asterisk-users Digest, Vol 51, Issue 48
i don`t want anymore this messages.thanks _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx___ -- 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] LDAP authentication
Hello I want to set up an LDAP authentication on my asterisk network by using the astirectory module. To avoid NAT problems, every client have a local asterisk server which is linked with the others by a main server. Firstly, I have installed he astirectory module on local server, but since it not only read in my LDAP but also write in it it cause me security problems. My question is: is there any way to configure my locals server to transfer the authentication request to the main one ? Thanks in advance. Adrien Couet ___ -- 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] mismatched callerid on phone and CDR ?
Hi, Using asterisk 1.4.21.2. For some calls (usally telemarketers) entering through a BRI zap channel I somtimes notice the callerid on my polycom 601 phone and the CDR's 'src' field don't match. They are even totally different. And the displayed callerid is nowhere to be seen in the CDR record. Is there a rational explanation? -- http://www.lesculturelles.net ___ -- 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] mismatched callerid on phone and CDR ?
On Wednesday 15 October 2008 10:26:50 Louis-David Mitterrand wrote: For some calls (usally telemarketers) entering through a BRI zap channel I somtimes notice the callerid on my polycom 601 phone and the CDR's 'src' field don't match. They are even totally different. And the displayed callerid is nowhere to be seen in the CDR record. Is there a rational explanation? The ANI and CallerID do not necessarily have to match; they just generally do. The src field reflects the ANI, if set, with a fallback to CallerID number, if not. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4
Hi John, this is getting quite strange, and i'm becoming quite curios why it's not working :) Could you try first setting up realtime for SIP or queues? This should work out-of-the-box on 1.4. http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue For beginning you can just add queues=.. in extconfig.conf to see that SQLs go trough and return errors. You should see SELECT's in your log whenever accessing. For example, entering into CLI: ast-dev14*CLI queue show myqueue would write into log: [Oct 15 04:04:09] DEBUG[9935] res_config_mysql.c: MySQL RealTime: Everything is fine. [Oct 15 04:04:09] DEBUG[9935] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM queue_table WHERE name = 'myqueue' Also i would suggest enabling full log, as it's one place you can see everything. Then use grep to search for realtime messages. Your logger.conf should already have commented line: full = notice,warning,error,debug,verbose Then you can do: # tail -fn0 /var/log/asterisk/full | grep -F res_config_mysql to see every message about realtime driver. After this you can unload and load module or restart asterisk completely (if restarting, make sure it's started with -vvvd). To reload module, use: ast-dev14*CLI module unload res_config_mysql.so MySQL RealTime unloaded. ast-dev14*CLI module load res_config_mysql.so == Parsing '/etc/asterisk/res_mysql.conf': Found MySQL RealTime driver loaded. Loaded res_config_mysql.so = (MySQL RealTime Configuration Driver) On Wed, Oct 15, 2008 at 9:05 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: Hi Atis, queue_log = mysql,asteriskcdrdb,queue_log that is engine,database,table If it's wrong, you should see some warnings when asterisk is starting up. Thanks for the suggestion. I did not put in queue_log for table and it has just taken the default which is queue_log. In the console startup, you can see below that it has successfully bound queue_log to /mysql/db1/queue_log. # asterisk -rvvv Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. [...] == Parsing '/etc/asterisk/extconfig.conf': Found == Binding queue_log to mysql/db1/queue_log Connected to Asterisk 1.4.21.2 currently running on machine Verbosity is at least 3 This seems somehow strange. If you connect to running asterisk with -r, you shouldn't see parsing extconfig.conf, as it should be parsed on startup time. You could also add -d to enable debug 1. In /var/log/asterisk/messages, I saw: [Oct 15 15:31:48] NOTICE[20941] config.c: Registered Config Engine mysql Another idea that came into my mind is, that (if this config doesn't still work) you might have to do make dist-clean within asterisk-addons after reinstalling asterisk, and then configure, make, make install. It's because addons do use headers from installed version of asterisk, and they might not have correct declarations. Basically, I did: - Asterisk-1.4.21.2 make clean ./configure make make install - Asterisk-addons-1.4.7 make dist-clean ./configure make make install Yes, this is completely correct (assuming you restarted asterisk after :) Also, you mentioned that you checked /var/log/asterisk/messages, however i think debug is written into file called debug. Anyway you can enable full in logger.conf and get everything there. To debug this you shouldn't need more than core set verbose 3 and core set debug 1. I turned on debug mode and tried an agent login and logoff. However, when I looked into debug and messages, there are lots of chan_sip.c and a few cdr_addon_mysql.c but no occurrence at all of res_config_mysql.c What is happening? Do I have to explicitly load it? *CLI module show like res_config_mysql Module Description Use Count res_config_mysql.so MySQL RealTime Configuration Driver 0 1 modules loaded This should also be fine. You could also try catching me on irc, just look for atis_work or atis_home in #asterisk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.21.2 - Issues with call parking
Hi, I am facing the following issues with call parking: 1) Asterisk does not announce the parking extension:- I press # and hear transfer followed by a beep. I dial 700 but do not hear the parking extension number. 2) After parking time, the original extension rings and i accept the call. When I again try to park, the call park sequence fails. I have attached features.conf and extensions.conf. The attached file capture.txt contains the Asterisk messages. Please let me know if I am missing something. Thanks, lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching *, + and # in the dialplan
On Tue, 2008-10-14 at 19:59 -0500, Karl Fife wrote: QUESTION: Is there a way to do just that? As in: match: one more of the preceding character or expression (a variation on '.') zero more of the preceding character or expression (a variation on bang) No, there's currently nothing in the Asterisk pattern matching syntax to constrain one digit to be related in any fashion to the preceding digit. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 1.4.21.2 - Issues with call parking
Muthukrishnan Venkatakrishnan wrote: Hi, I am facing the following issues with call parking: 1) Asterisk does not announce the parking extension:- I press # and hear transfer followed by a beep. I dial 700 but do not hear the parking extension number. You need to enable one touch parking instead of manually sending a call to 700. According to your features.conf, you have blind transfer setup to #1. If you want to transfer to 700, you need to be using attended transfers, not blind. 2) After parking time, the original extension rings and i accept the call. When I again try to park, the call park sequence fails. There is a patch available that was posted just a few days back. Search the archives against my name. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 1.4.21.2 - Issues with call parking
Forgot the attachments features.conf Description: Binary data extensions.conf Description: Binary data -- Executing [EMAIL PROTECTED]:1] Dial(SIP/lee-00168b18, SIP/jeya||tTkK) in new stack -- Called jeya -- SIP/jeya-0016a0a8 is ringing -- SIP/jeya-0016a0a8 answered SIP/lee-00168b18 -- Started music on hold, class 'default', on SIP/lee-00168b18 -- SIP/jeya-0016a0a8 Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/lee-00168b18 -- Started music on hold, class 'default', on SIP/lee-00168b18 == Parked SIP/lee-00168b18 on [EMAIL PROTECTED] Will timeout back to extension [local] 300, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls == Spawn extension (local, 300, 1) exited KEEPALIVE on 'SIP/lee-00168b18' -- Stopped music on hold on SIP/lee-00168b18 -- Registered extension context 'park-dial' -- Added extension 'SIP/jeya' priority 1 to park-dial == Timeout for SIP/lee-00168b18 parked on 701. Returning to park-dial,SIP/jeya,1 -- Executing [SIP/[EMAIL PROTECTED]:1] Dial(SIP/lee-00168b18, SIP/jeya|30|t) in new stack -- Called jeya -- SIP/jeya-0016f818 is ringing -- SIP/jeya-0016f818 answered SIP/lee-00168b18 -- Started music on hold, class 'default', on SIP/lee-00168b18 -- SIP/jeya-0016f818 Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on SIP/lee-00168b18 ___ -- 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] Help With AMI
The Asterisk-Java project has a working API implementation that includes this command, and has it documented here: http://asterisk-java.org/development/apidocs/org/asteriskjava/manager/ac tion/UpdateConfigAction.html It's sort of an all-in-one action. Scroll down in the link above to the addCommand method, and there's some explanation of the fields. Hope that helps, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Dickenson Sent: Tuesday, October 14, 2008 6:58 PM To: Asterisk User MailList Subject: [asterisk-users] Help With AMI I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+ API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-00:append Cat-00:newuser Var-00:secret Value-00:nottelling I have searched various web sites and mail lists but I can not find very much documentation about how the updateconfig action is to work. Can anyone point me to additional documentation in addition to offering some ideas as to why the above transaction might not work. TIA -- Jim Dickenson mailto:[EMAIL PROTECTED] CfMC http://www.cfmc.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk+heartbeat
How did you define the secondary IP address? Did you actually set that up in the network scripts and bind it to eth0 or did you just define it in /etc/ha.d/haresources? You should only have the virtual IP defined in haresources along with the primary server and what you want to do on node up/down. My haresources file has a single line: ohasterisk01 10.191.32.31 MailTo::user@domain.com::Asterisk fonulator asterisk We're obviously using the redFone FoneBRIDGE for our T1 connection as you can see we're firing the fonulator script and then asterisk. HTH! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nhadie Sent: Tuesday, October 14, 2008 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk+heartbeat Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 eth0 10.10.10.3 secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie ___ -- 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] Zaptel compile error after make update.
Hi, I started to get some Zaptel compile errors after a 'make update' I did a clean zaptel install with: svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel I am still getting the error, is this someelse seeing this ?. CC [M] /usr/src/zaptel/kernel/zaptel-base.o /usr/src/zaptel/kernel/zaptel-base.c: In function 'zt_reallocbufs': /usr/src/zaptel/kernel/zaptel-base.c:889: error: 'struct zt_chan' has no member named 'rebufpolicy' make[3]: *** [/usr/src/zaptel/kernel/zaptel-base.o] Error 1 make[2]: *** [_module_/usr/src/zaptel/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.23.15-80.fc7-i686' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/zaptel' make: *** [all] Error 2 It's a FC7 and the Zaptel cards is a TE410P Freddi ___ -- 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] phoniceq e400p driver for DAHDI
Hello everyone, We have an E400P Card from phoniceq. There is a DAHDI Driver posted at: http://e400p.phoniceq.com/driver/dahdi-tor2-tormenta3-e1.tgz but, it doesn't work. The author (martin (or marcin) pycko) says that it isn't finished. I've e-mailed martin, and he stated that he would fix the driver AFTER we order 10 more cards, not before. I have a difficult time ordering thousands of dollars worth of product BEFORE i see it working correctly. During my life, I've been promised a lot of things If anyone can fix the driver, I'll gladly pay a reasonable fee for the service. The hardware seems to work well, and it definately fits my budget, but i'm more comfortable paying a few dollars to get this done BEFORE i buy the cards. Cheap insurance. To replicate my setup, you would need to download dahdi-linux-complete-2.0.0.tar.gz , untar it, and then download the file above and copy the contents over the top of .../dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi/tor2.c Below is the debug information I provided when I first contacted phoniceq. If anyone here thinks they can tackle this for me, Please get in touch via direct email and let me know how much you want to fix it up. Thanks Everyone, Josh --- When I insmod/modprobe tor2.c, however, I get a segmentation fault, and I can't use the driver, or even unload it. The only way to remove the driver is to reboot the machine. develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # uname -a Linux develop 2.6.25.5-1.1-pae #1 SMP 2008-06-07 01:55:22 +0200 i686 i686 i386 GNU/Linux develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # insmod tor2.ko Segmentation fault develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # rmmod tor2 ERROR: Module tor2 is in use develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # *NOTE*(tor2 is NOT in use)**NOTE* develop:/usr/src/tor2/dahdi-linux-complete-2.0.0+2.0.0/linux/drivers/dahdi # dahdi_cfg -vvv DAHDI Tools Version - 2.0.0 DAHDI Version: 2.0.0 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) Channel 32: Clear channel (Default) (Slaves: 32) Channel 33: Clear channel (Default) (Slaves: 33) Channel 34: Clear channel (Default) (Slaves: 34) Channel 35: Clear channel (Default) (Slaves: 35) Channel 36: Clear channel (Default) (Slaves: 36) Channel 37: Clear channel (Default) (Slaves: 37) Channel 38: Clear channel (Default) (Slaves: 38) Channel 39: Clear channel (Default) (Slaves: 39) Channel 40: Clear channel (Default) (Slaves: 40) Channel 41: Clear channel (Default) (Slaves: 41) Channel 42: Clear channel (Default) (Slaves: 42) Channel 43: Clear channel (Default) (Slaves: 43) Channel 44: Clear channel (Default) (Slaves: 44) Channel 45: Clear channel (Default) (Slaves: 45) Channel 46: Clear channel (Default) (Slaves: 46) Channel 47: Clear channel (Default) (Slaves: 47) Channel 48: D-channel (Default) (Slaves: 48) Channel 49: Clear channel (Default) (Slaves: 49) Channel 50: Clear channel (Default) (Slaves: 50)
Re: [asterisk-users] Matching *, + and # in the dialplan
On Wed, 2008-10-15 at 09:06 -0400, Jared Smith wrote:
On Tue, 2008-10-14 at 19:59 -0500, Karl Fife wrote:
QUESTION: Is there a way to do just that? As in: match:
one more of the preceding character or expression (a variation on '.')
zero more of the preceding character or expression (a variation on bang)
No, there's currently nothing in the Asterisk pattern matching syntax to
constrain one digit to be related in any fashion to the preceding digit.
Jared is correct.
What you really want is the RE *, +, and maybe even () features.
Not to mention '?'...
Some RE features would be easy to implement in the trie, but
the real killer is trailing context... for instance...
XX[58]*ZZ
If you give it the pattern 3358, it has to decide that
the [58]* part is empty and the 58 is matched by ZZ.
And this makes the whole algorithm pretty hairy.
The current notation lends itself to a fast left-to-right
evaluation, without multiple recursive attempts to find
a path that would lead to a match.
But, if you are willing to forego trailing context,
and make it so any *,+, {x,z}, or ? is at the end of an expression,
like . is now, this could be implemented fairly straightforwardly in our
current pattern matchers.
See my previous conversations in the dev mailing list, back in aug
2007...
let's see:
http://lists.digium.com/pipermail/asterisk-dev/2007-August/028844.html
http://lists.digium.com/pipermail/asterisk-dev/2007-August/028846.html
http://lists.digium.com/pipermail/asterisk-dev/2007-August/028858.html
murf
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Steve Murphy
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Re: [asterisk-users] SIP channels seem not to close after call is finished
On Tue, 2008-10-14 at 17:24 -0500, Daniel - Asterisk wrote: Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [EMAIL PROTECTED] asterisk -rx sip show channels |grep 648 Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.25.29648 7c24869b010 00102/0 0x2 (gsm) No Tx: ACK 192.168.25.29648 26e8187a0a4 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 5289c52b77e 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 7a6243bc21e 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 32bcf3ea3f9 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 21ff7be5355 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 04725bda23e 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 2e9a9db559c 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 7fab5e8044d 00102/0 0x0 (nothing) No Tx: CANCEL 192.168.25.29648 11313fc173a 00102/0 0x0 (nothing) No Tx: CANCEL 2. Asterisk version: 1.4.21.1 These look a lot like the Zombie Channel Bloating Death problems we attacked over the last few weeks. Please see if the latest svn version of 1.4 has these problems still. In high-volume systems, this looked like a huge memory leak that would lead to death by swiftly using up memory, file descriptors, etc. until Asterisk ran out of virtual memory and crashed. There are a couple of code paths, one leaves CANCELED channels lying around, the other BYE'd channels. murf 3. I'm using SIP realtime peers, sip.conf configuration follows: [general] bindport=5060 bindaddr=0.0.0.0 context=default language=es rtcachefriends=yes disallow=all allow=ulaw allow=alaw allow=gsm rtpholdtimeout=300 rtptimeout=300 dtmfmode=rfc2833 videosupport=yes progressinband=yes allowsubscribe=yes subscribecontext=extensiones notifyringing=yes notifyhold= yes limitonpeers= yes Daniel Arohuanca Lagos +51 1 994149553 Lima-Peru ___ -- 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 -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel compile error after make update.
On Wed, Oct 15, 2008 at 09:38:17PM +0200, Freddi Hansen wrote: Hi, I started to get some Zaptel compile errors after a 'make update' I did a clean zaptel install with: svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel I am still getting the error, is this someelse seeing this ?. CC [M] /usr/src/zaptel/kernel/zaptel-base.o /usr/src/zaptel/kernel/zaptel-base.c: In function 'zt_reallocbufs': /usr/src/zaptel/kernel/zaptel-base.c:889: error: 'struct zt_chan' has no member named 'rebufpolicy' It looks like a typo from a recent commit. Try replacing 'rebufpolicy' with: 'rxbufpolicy' in line 899. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail.conf
Is it possible to create extensions in the voicemail.conf remotely by using the manager interface. I cannot seem to find any documents or examples describing that capability. Jonathan ___ -- 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] voicemail.conf
On Wednesday 15 October 2008 15:59:09 jonathan augenstine wrote: Is it possible to create extensions in the voicemail.conf remotely by using the manager interface. I cannot seem to find any documents or examples describing that capability. It's not possible, no. However, you could enable realtime voicemail and configure new voicemail users via a database. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help With AMI
I have figured out what I was doing wrong. Although when I issued my action I got a good return there were problems with my action. One thing I needed to do was a newcat before I adding values. Would have been nice if I had gotten some error saying trying to add values to a non-existant cat. In any event I have updated both the agent.conf and manager.conf files. Will try others to see how far I get. Is there any more documentation other than reading the source code and this little bit here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig -- Jim Dickenson mailto:[EMAIL PROTECTED] CfMC http://www.cfmc.com/ From: Martin Smith [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 15 Oct 2008 09:09:48 -0400 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Conversation: [asterisk-users] Help With AMI Subject: Re: [asterisk-users] Help With AMI The Asterisk-Java project has a working API implementation that includes this command, and has it documented here: http://asterisk-java.org/development/apidocs/org/asteriskjava/manager/ac tion/UpdateConfigAction.html It's sort of an all-in-one action. Scroll down in the link above to the addCommand method, and there's some explanation of the fields. Hope that helps, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Dickenson Sent: Tuesday, October 14, 2008 6:58 PM To: Asterisk User MailList Subject: [asterisk-users] Help With AMI I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+ API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-00:append Cat-00:newuser Var-00:secret Value-00:nottelling I have searched various web sites and mail lists but I can not find very much documentation about how the updateconfig action is to work. Can anyone point me to additional documentation in addition to offering some ideas as to why the above transaction might not work. TIA -- Jim Dickenson mailto:[EMAIL PROTECTED] CfMC http://www.cfmc.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Configuring SIP TLS
Hi friends, I need a help to configure the TLS certificate chains to use with Asterisk for SIP TLS, can anyone help me with this, sending a link, a tutorial or something like that which explains how to generate and use CAs??? -- Thanks, Rafael Puga ___ -- 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+heartbeat
I had an issue in 1.2.x that I am not sure ever got fixed but having two NICs on the same subnet did strange things. Strange like one NIC would be used for outbound traffic and the other for inbound traffic. Maybe someone knows why, if it was a bug, if it was fixed? I could think of something creative but am drained right now and have to go buy some coax and connectors. If nobody helps you by then, I will. Thanks, Steve Totaro On Wed, Oct 15, 2008 at 3:39 PM, Gleim, Jason [EMAIL PROTECTED]wrote: How did you define the secondary IP address? Did you actually set that up in the network scripts and bind it to eth0 or did you just define it in /etc/ha.d/haresources? You should only have the virtual IP defined in haresources along with the primary server and what you want to do on node up/down. My haresources file has a single line: ohasterisk01 10.191.32.31 MailTo::user@domain.com::Asterisk fonulator asterisk We're obviously using the redFone FoneBRIDGE for our T1 connection as you can see we're firing the fonulator script and then asterisk. HTH! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nhadie Sent: Tuesday, October 14, 2008 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk+heartbeat Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 eth0 10.10.10.3 secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie ___ -- 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+heartbeat
Are you using bindaddr=0.0.0.0 in sip.conf and iax.conf ?? On Wed, Oct 15, 2008 at 5:35 PM, Steve Totaro [EMAIL PROTECTED] wrote: I had an issue in 1.2.x that I am not sure ever got fixed but having two NICs on the same subnet did strange things. Strange like one NIC would be used for outbound traffic and the other for inbound traffic. Maybe someone knows why, if it was a bug, if it was fixed? I could think of something creative but am drained right now and have to go buy some coax and connectors. If nobody helps you by then, I will. Thanks, Steve Totaro On Wed, Oct 15, 2008 at 3:39 PM, Gleim, Jason [EMAIL PROTECTED]wrote: How did you define the secondary IP address? Did you actually set that up in the network scripts and bind it to eth0 or did you just define it in /etc/ha.d/haresources? You should only have the virtual IP defined in haresources along with the primary server and what you want to do on node up/down. My haresources file has a single line: ohasterisk01 10.191.32.31 MailTo::user@domain.com::Asterisk fonulator asterisk We're obviously using the redFone FoneBRIDGE for our T1 connection as you can see we're firing the fonulator script and then asterisk. HTH! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nhadie Sent: Tuesday, October 14, 2008 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk+heartbeat Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 eth0 10.10.10.3 secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie ___ -- 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] Cisco 7960 not always receiving incoming calls
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest phone image from Cisco to date.
I am also using a Cisco router at the gateway. Is there anything
special I should to to make this work? Note my soft phone does not
have any issues using the same dialing rules and extension
information. Here is some of my config stuff:
ns1*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
vitel-inbound/rsreese 64.2.142.1165060 Unmonitored
101/10168.156.63.118D N 1038 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]
Inbound call in progress when the SIP Cisco phone doesn't ring
Verbosity is at least 5
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on 'SIP/rsreese-082a8358'
Inbound call in progress when the SIP Cisco does ring after I first
make an outbound call
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on 'SIP/rsreese-082a8358'
Extensions.conf, which I don't think is relevent, I've changed it to
just a simple dial the sip phone and it still fails.
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
Cisco phone stuff from a Cisco 7960:
SIPDefault.cnf
image_version: P0S3-08-9-00
proxy1_address: neocipher.net; Can be dotted IP or FQDN
proxy_register: 1
messages_uri: 100
phone_password: cisco ; Limited to 31 characters (Default - cisco)
sntp_server:10.10.10.1
time_zone: EST
dial_template: DIALPLAN
nat_enable: 1
nat_address: 172.16.2.1
nat_received_processing: 1
outbound_proxy_port: 5060
outbond_proxy: ns1.neocipher.net
SIP0112B9EAFF72.cnf
image_version: P0S3-08-9-00
# Line 1 Setup
line1_name: 101
line1_authname: 101
line1_shortname: Line 101
line1_password: test
line1_displayname: Stephen Reese; # Line 1 Display Name (Display
name to use for SIP messaging)
# Line 2 Setup
#line2_name: scott
#line2_authname: scott
#line2_shortname: 201
#line2_password: tiger
#line2_displayname: Larry Ellison; # Line 2 Display Name (Display
name to use for SIP messaging)
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: Stephen Reese ; Has no effect on SIP messaging
# Phone Password (Password to be used for console or telnet login)
phone_password: goaway ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
telnet_level: 2
Any ideas or help would be great, thanks.
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[asterisk-users] Queue problem
Hi, I have 3 queues and they have the same weight. But one of the queues receives a lot of calls (much more than the other two) so people on that queue usually have to wait much more than the others. Is there a way to make asterisk determine the longest waiting call and give priority to that call, having the 3 queues (I know that if I had just one queue, this would be the natural behavior). Thanks, Jorge Santiago Alanís Garza Innovación y Desarrollo mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Tel: (81) .4044 Cel: (811) 243-6570 http://www.blocknetworks.com.mx/ www.blocknetworks.com.mx Av. Lázaro Cárdenas 4000, L-17 Col. Valle de las Brisas Monterrey, Nuevo León, CP 64790 Tel: +52 (81) 4044 Block Networks es una empresa certificada en ISO 9001:2000 Design, development, and sales of enterprise software and technology. La información contenida en este mensaje y sus anexos es de carácter privado y confidencial y para el uso exclusivo de la persona o institución a la cual ha sido enviado y para otros autorizados para recibirlo, por lo que no podrá distribuirse sin la autorización expresa del remitente. Si usted no es el destinatario a quien este mensaje fue dirigido o si no es un empleado responsable del envío de este mensaje al destinatario, se hace de su conocimiento que cualquier revisión, diseminación, distribución, copia u otro uso o acto realizado con base en o relacionado con el contenido de este mensaje y sus anexos está estrictamente prohibida y puede ser ilegal. Asimismo, el presente mensaje no representa la manifestación del consentimiento de ninguna de las partes, por lo que no genera derecho u obligación alguna para ambas sino hasta que sus representantes legales así lo manifiesten por escrito. Si usted ha recibido este comunicado y sus anexos por error, le solicitamos lo notifique inmediatamente al remitente respondiendo a este correo y borre el presente y sus anexos de su sistema sin conservar copia de los mismos. Gracias, Block Networks, S.A. de C.V. The information contained in this message and its attachments is private and confidential and is intended solely for the use of the individual or entity to whom it is addressed and others who are authorized to receive it; therefore, its distribution cannot be possible without authorization from the sender. If you are not the intended recipient or an employee responsible for delivering this message to the intended recipient, you are hereby notified that any revision, dissemination, distribution, copying or other use or action based upon or relative to the information contained in this message and its attachments is strictly prohibited and may be unlawful. You are also informed that the contents of this message shall not be considered as an agreement between the parties and shall not bind any of them until their attorneys decide to do so in writing. If you have received this message and its attachments by error, please immediately notify the sender by replying to this message and deleting it from your system without keeping a copy. Thank you. Block Networks, S.A. de C.V. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
Being a Panasonic dealer and having more than 50 Asterisk system in production, I can tell you that if this is your first Asterisk project, then go with Panasonic, you'll safe yourself lots of aggravation and have a happier customer. Some features of the Panasonic you will never be able to emulate on Asterisk. While depending on the needs of that customer, and in some cases I would suggest dive into Asterisk, I gather from the subject (yes I have read the whole message, for those of you out there that might think that I did not) that a Panasonic will work nicely for them, therefore my advice stick with Panasonic. On Mon, Oct 13, 2008 at 9:15 PM, Rodolfo Alcazar Portillo [EMAIL PROTECTED] wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. * I need to get a formatted calls report for the administrators to charge the users. I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. Thanks a lot. -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!
On Mon, Oct 13, 2008 at 11:54 PM, Jorge Mendoza [EMAIL PROTECTED] wrote: Rodolfo Alcazar Portillo wrote: Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. Asterisk is more featured than Panasonic, but you must to know Asterisk to convince your executives ;-) Not really so. Depending on lots of factors, usually for a small office of only 5-10 users Panasonic is more feature rich. Since the main feature they are looking for in a PBX is to be able to yell across the hallway; hey boss call on 5 it's your wife which is not really possible with Asterisk (yeah I know call parking, but how many phones support it flawlessly with flashing LEDs?). Other features that are quite popular in small offices and not supported by Asterisk: * Live call screening - Yes there is a hack that can do it, but it's a hell of a hack. * Phones that can do most of the usefull features supported by the PBX for a reasonable price with LED buttons, including the following features: ** Call recording with LED indication, while at it, the recordings integrate seamlessly with your voicemail, which means you don't need to browse the file system on the PBX to listen to it. ** Login/Logout of queues, Day/Night mode buttons with indication (1.6 has this as well). ** Company internal directory on the phone updated on the PBX ** System Speed Dial on the display updated by the PBX ** Call Fwd by PBX with LED indication (not phone based callfwd which sucks). ** On screen Voicemail (on the phone). ** Line assignment to buttons with LED indication, and hold indication. ** Hold ringback (some IP phones support it). There are many more features but I can't remember them at the moment. Granted in bigger installations there many more factors and usually more funding which makes the above list almost obsolete for the features that Asterisk does have. Again my advice do not go with Asterisk for this installation go with Panasonic. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP-PSTN and PSTN-SIP calls. Works. Now, I need this help, please: * Dialing from inside (pap2-FXS connected phone) to another number on the same city (goes out by SPA3102 FXO), voice works fine. But when a menu answers, and I dial over, the menu dialed keys works only 20% of all times. Why could this would be? Voltage levels? sound gains? Dialed keys get distorsioned when passing over the 2 Linksys? Linksys or Asterisk swallowing some dialed key? I noticed some echo... Probably you are sending dtmf signals inband. Try outband. For the echo, try to change the FXO/FXS impedance, and/or playing with the rx and tx gains. I assume that do you have echo cancelling enable in both SPA. * I need to assign two codes to each user, one for international calls charged to the office, another for international calls charged to the user. If the user enters an incorrect code, the call should not proceed. See account codes. You can start here: http://www.voip-info.org/wiki-Asterisk+Billing * I need to get a formatted calls report for the administrators to charge the users. See same link, or google for billing I just am confused and stucked with all the documentation in Internet, and all this new asterisk jargon. I just need some links (or some directions) to go fast on this topics. Of course, some more help would be appreciated. The link to start: http://www.voip-info.org Thanks a lot. De nada Jorge ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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+heartbeat
hi sir as mentioned i need to bind it on the secondary ip. coz if i bind it to 0.0.0.0, it's rejecting the call. i'm not sure if possible with heartbeat to execute a command after it takes over an IP. so i can reload asterisk on the failover server once it has the ip so i can bind asterisk to it. regards nhadie Edgar Guadamuz wrote: Are you using bindaddr=0.0.0.0 http://0.0.0.0 in sip.conf and iax.conf ?? On Wed, Oct 15, 2008 at 5:35 PM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I had an issue in 1.2.x that I am not sure ever got fixed but having two NICs on the same subnet did strange things. Strange like one NIC would be used for outbound traffic and the other for inbound traffic. Maybe someone knows why, if it was a bug, if it was fixed? I could think of something creative but am drained right now and have to go buy some coax and connectors. If nobody helps you by then, I will. Thanks, Steve Totaro On Wed, Oct 15, 2008 at 3:39 PM, Gleim, Jason [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: How did you define the secondary IP address? Did you actually set that up in the network scripts and bind it to eth0 or did you just define it in /etc/ha.d/haresources? You should only have the virtual IP defined in haresources along with the primary server and what you want to do on node up/down. My haresources file has a single line: ohasterisk01 10.191.32.31 http://10.191.32.31/ MailTo::user@domain.com::Asterisk fonulator asterisk We're obviously using the redFone FoneBRIDGE for our T1 connection as you can see we're firing the fonulator script and then asterisk. HTH! Jason -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:asterisk-users- mailto:asterisk-users- [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Nhadie Sent: Tuesday, October 14, 2008 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk+heartbeat Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 http://10.10.10.1/ eth0 10.10.10.3 http://10.10.10.3/ secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 http://10.10.10.2/ eth0 10.10.10.3 http://10.10.10.3/ secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk+heartbeat
Gleim, Jason wrote: How did you define the secondary IP address? Did you actually set that up in the network scripts and bind it to eth0 or did you just define it in /etc/ha.d/haresources? You should only have the virtual IP defined in haresources along with the primary server and what you want to do on node up/down. hi sir, secondary is asisgned by heartbeat via haresources. My haresources file has a single line: ohasterisk01 10.191.32.31 MailTo::user@domain.com::Asterisk fonulator asterisk does this mean asterisk will run after it takes over the IP? can i do this without the fonulator script? We're obviously using the redFone FoneBRIDGE for our T1 connection as you can see we're firing the fonulator script and then asterisk. HTH! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nhadie Sent: Tuesday, October 14, 2008 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk+heartbeat Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 eth0 10.10.10.3 secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello, On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: A packet trace will probably show exactly what is happening. Try: tcpdump -nlXs 8192 -i eth0 port 5060 You should be able to see the SIP information going back and forth and will probably show you that your NAT rules are applying when they shouldn't. I agree with first turning off your firewall and testing... but if that actually solves the problem you need to know why. This should tell why. Why eth0 when in fact it is not being used AFAIK? My Asterisk box is connected to the LAN via its eth1 interface and the SIP phone is calling from the LAN to the analog telephone via FXO/POTS. Again, below is the call scenario diagram: [SNOM] ==LAN== eth1 [ASTERISK] fxo ==POTS== [ANALOG_TELEPHONE] eth0 || INTERNET Please advice. Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Daniel, On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only worked one way. I ended up rigging * to force the line to be considered answered with a patch. Yes, the call has been answered as per Asterisk logs. The CALLER (SNOM SIP Phone) can hear clearly the voice of the target CALLEE (POTS analog telephone) but it is the CALLEE that cannot hear the CALLER's voice. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching *, + and # in the dialplan
On Wed, 15 Oct 2008 14:22:09 -0600, Steve Murphy [EMAIL PROTECTED]
said:
the real killer is trailing context... for instance...
XX[58]*ZZ
If you give it the pattern 3358, it has to decide that
the [58]* part is empty and the 58 is matched by ZZ.
And this makes the whole algorithm pretty hairy.
The current notation lends itself to a fast left-to-right
evaluation, without multiple recursive attempts to find
a path that would lead to a match.
But, if you are willing to forego trailing context,
and make it so any *,+, {x,z}, or ? is at the end of an expression,
like . is now, this could be implemented fairly straightforwardly in our
current pattern matchers.
So how would one route calls differently if they're ISN formatted i.e.
'6565*696'. I can't get my head around any way to do that using the
existing rules.
Freenum.org suggests an ISN Prefix such as _012. to disambiguate ISN's
this is a total kludge because ISN-formatted numbers are already
perfectly unambiguous (not to mention the obvious limitation that
DIALING an ISN from a given system would first involve an query to the
admin).
The obvious problem is that the disambiguating character is located
anywhere between the second and fourth-from-last character. (one or more
digits followed by *, followed by three or more digits). Cursed
trailing contexts!
The only thing I can think of is to categorically exclude it from all
other possiblities: for example:
_NXX local number
_1NXXNXX non-local number
_011XX. international
_*XX supplemental service codes
_+XX. international
_X. ISN ???
...if that would even work.
Is there a better way to do this?
Is how 'expensive' would it be to disambiguate based on the unique
characteristics of an ISN?
It seems like there should be a inexpensive, non-recursive, one-pass way
to do it, but without getting my head inside the parser like Steve
has...
Thanks!
-Karl
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Re: [asterisk-users] asterisk+heartbeat
Version 2 of Linux-HA allows order constraints, so you can first set the IP and then start asterisk. (service asterisk should be down until IP address moves) On Wed, Oct 15, 2008 at 7:17 PM, Nhadie [EMAIL PROTECTED] wrote: Gleim, Jason wrote: How did you define the secondary IP address? Did you actually set that up in the network scripts and bind it to eth0 or did you just define it in /etc/ha.d/haresources? You should only have the virtual IP defined in haresources along with the primary server and what you want to do on node up/down. hi sir, secondary is asisgned by heartbeat via haresources. My haresources file has a single line: ohasterisk01 10.191.32.31 MailTo::user@domain.com::Asterisk fonulator asterisk does this mean asterisk will run after it takes over the IP? can i do this without the fonulator script? We're obviously using the redFone FoneBRIDGE for our T1 connection as you can see we're firing the fonulator script and then asterisk. HTH! Jason -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nhadie Sent: Tuesday, October 14, 2008 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk+heartbeat Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 eth0 10.10.10.3 secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk+heartbeat
So Redfone relies on http running? That is almost as bad as using ping. HTTP could be up but Asterisk could have crapped out but ping and HTTP could still be up. I think the manager is the best way. If your backup box has 10.10.10.3 hardcoded and up, you have an IP conflict and I am surprised it works at all. Just a thought, but how about you run a cron job on your backup that connects to your primary Asterisk boxes' AMI and issues some benign command and pings it. If it works as expected, then all is well, if it AMI doesn't reply, continue your cron to ping. If no ping replies, then proceed to bring up 10.10.10.3. on your secondary box If ping works but asterisk doesn't, you could continue your cron job to ssh the primary box and disable the 10.10.10.3 NIC and then bring up the 10.10.10.3 NIC on the spare. If this is for SIP, just use OpenSer, if for TDM, use Redfone. Thanks, Steve Totaro On Wed, Oct 15, 2008 at 9:12 PM, Nhadie [EMAIL PROTECTED] wrote: hi sir as mentioned i need to bind it on the secondary ip. coz if i bind it to 0.0.0.0, it's rejecting the call. i'm not sure if possible with heartbeat to execute a command after it takes over an IP. so i can reload asterisk on the failover server once it has the ip so i can bind asterisk to it. regards nhadie Edgar Guadamuz wrote: Are you using bindaddr=0.0.0.0 http://0.0.0.0 in sip.conf and iax.conf ?? On Wed, Oct 15, 2008 at 5:35 PM, Steve Totaro [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I had an issue in 1.2.x that I am not sure ever got fixed but having two NICs on the same subnet did strange things. Strange like one NIC would be used for outbound traffic and the other for inbound traffic. Maybe someone knows why, if it was a bug, if it was fixed? I could think of something creative but am drained right now and have to go buy some coax and connectors. If nobody helps you by then, I will. Thanks, Steve Totaro On Wed, Oct 15, 2008 at 3:39 PM, Gleim, Jason [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: How did you define the secondary IP address? Did you actually set that up in the network scripts and bind it to eth0 or did you just define it in /etc/ha.d/haresources? You should only have the virtual IP defined in haresources along with the primary server and what you want to do on node up/down. My haresources file has a single line: ohasterisk01 10.191.32.31 http://10.191.32.31/ MailTo::user@domain.com::Asterisk fonulator asterisk We're obviously using the redFone FoneBRIDGE for our T1 connection as you can see we're firing the fonulator script and then asterisk. HTH! Jason -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:asterisk-users- mailto:asterisk-users- [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Nhadie Sent: Tuesday, October 14, 2008 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk+heartbeat Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 http://10.10.10.1/ eth0 10.10.10.3 http://10.10.10.3/ secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 http://10.10.10.2/ eth0 10.10.10.3 http://10.10.10.3/ secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
[asterisk-users] Telrad Analog CID
Does anyone know if I have an older Telrad PBX if I can get CallerID to Asterisk when the connection is via analog FXO-FXS? I only need 1 or 2 lines so T1 is an overkill. ___ -- 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] How to invoke an external C program and output an integer to the program?
Hi, I want to call an extension like 8 and invoke an external C program upon calling, pass an constant integer like 1 to the C program. What I have done is: /etc/extensions.conf: exten = 8,1,system(/usr/local/src/parallel/fire 1) exten = 8,n, Dial(SIP/8) exten = 8,n,Hangup the C program under /usr/local/src/parallel/fire will wait for the input, if it's 1 external LED light will be on, if it's 0 LED light will be off. I have changed the file ownership and group since my asterisk user is asterisk (with freepbx): [EMAIL PROTECTED] parallel]# ls -l fire* -rwxrwxrwx 1 asterisk asterisk 5882 Oct 16 09:18 fire -rw-rw-rw- 1 asterisk asterisk 2793 Oct 15 22:25 fire.c If I run the program separately everything is fine: [EMAIL PROTECTED] parallel]# /usr/local/src/parallel/fire 1 buffer is 1 input1 value is 1 open port successfully Input1 is high, Pin 17 set to high Input1 is 1 [EMAIL PROTECTED] parallel]# /usr/local/src/parallel/fire 0 buffer is 0 input1 value is 0 open port successfully Input1 is low, Pin 17 set to low Input1 is 0 [EMAIL PROTECTED] parallel]# However if I call to 8 I can see it execute the system command but it doesn't output an integer 1 to my 'fire' program. CLI: -- Executing System(SIP/10-09a63138, /usr/local/src/parallel/fire 1) in new stack Any ideas on this or I shouldn't use System() at all? Thanks Best Regards, Johnny Xing ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Did you try it the magic number of times, three? On Sun, Oct 12, 2008 at 9:57 PM, GNUbie [EMAIL PROTECTED] wrote: Hello Tzafrir, On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: This means Zaptel gets silence from Asterisk. What codecs are used? What do you see on 'sip show channels'? I am using the following codecs: # asterisk -rx 'sip show settings' | grep Codecs Codecs: 0xe (gsm|ulaw|alaw) Below is the CLI output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/102-081d11d0, Zap/4/1234567) in new stack -- Called 4/1234567 *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 192.168.101.102 102 3c27a6824ba 00101/2 0x4 (ulaw) No Rx: INVITE 1 active SIP channel *CLI core show channels Channel Location State Application(Data) Zap/4-1 [EMAIL PROTECTED] Dialing AppDial((Outgoing Line)) SIP/102-081d11d0 [EMAIL PROTECTED]:1 RingDial(Zap/4/1234567) 2 active channels 1 active call Can you call from the FXO to Asterisk? (e.g.: to echo test) There is no problem with an inbound calls. I just tried to call the echo test extension number from my mobile phone via FXO/POTS and it works fine. I can hear my own voice. Thank you. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Change all canreinvites to no. On Wed, Oct 15, 2008 at 9:37 PM, GNUbie [EMAIL PROTECTED] wrote: Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk+heartbeat
Edgar Guadamuz wrote: Version 2 of Linux-HA allows order constraints, so you can first set the IP and then start asterisk. (service asterisk should be down until IP address moves) hi sir, i think this should do the trick for me. asterisk should be stopped, then when it gets the VIP, i will start asterisk and bind it to the VIP. is this how i should do it? asterisk-2 \ LVSSyncDaemonSwap::master \ IPaddr2::10.10.10.3/28/eth1/10.10.10.15 asterisk TIA. Regards, Ron On Wed, Oct 15, 2008 at 7:17 PM, Nhadie [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Gleim, Jason wrote: How did you define the secondary IP address? Did you actually set that up in the network scripts and bind it to eth0 or did you just define it in /etc/ha.d/haresources? You should only have the virtual IP defined in haresources along with the primary server and what you want to do on node up/down. hi sir, secondary is asisgned by heartbeat via haresources. My haresources file has a single line: ohasterisk01 10.191.32.31 http://10.191.32.31/ MailTo::user@domain.com::Asterisk fonulator asterisk does this mean asterisk will run after it takes over the IP? can i do this without the fonulator script? We're obviously using the redFone FoneBRIDGE for our T1 connection as you can see we're firing the fonulator script and then asterisk. HTH! Jason -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:asterisk-users- mailto:asterisk-users- [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Nhadie Sent: Tuesday, October 14, 2008 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk+heartbeat Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 http://10.10.10.1/ eth0 10.10.10.3 http://10.10.10.3/ secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 http://10.10.10.2/ eth0 10.10.10.3 http://10.10.10.3/ secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 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 http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
canreinvite defaults to yes, whether specified or not. http://www.voip-info.org/wiki/view/tips If you follow these directions adapting to your particular circumstances and it doesn't work, post your whole sip.conf Start asterisk with verbose set to 3 or so and turn on sip debugging. I get somewhere in the debug, you will see local NAT IPs that don't belong there, or it will just work. Thanks, Steve Totaro On Thu, Oct 16, 2008 at 12:12 AM, Steve Totaro [EMAIL PROTECTED] wrote: Change all canreinvites to no. On Wed, Oct 15, 2008 at 9:37 PM, GNUbie [EMAIL PROTECTED] wrote: Hello Karsten, On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote: Please post Your sip.conf. Which IP-Address do You configure in the snom for Your asterisk? (eth0 or eth1)? The SNOM 300 is using the NET interface beside the DC 5V port to connect to the LAN. The Asterisk box is using the eth1 to connect to the LAN. As per your instruction, below is my /etc/asterisk/sip.conf : - - - s n i p - - - [general] realm=pbx.domain.com bindport=5060 bindaddr=0.0.0.0 rtptimeout=60 disallow=all allow=ulaw allow=alaw allow=gsm externip=pbx.domain.com localnet=192.168.101.0/255.255.255.0 jbforce=yes allowtransfers=yes maxexpiry=3600 minexpiry=1800 videosupport=no [internal-phones](!) type=friend host=dynamic context=family dtmfmode=rfc2833 insecure=port,invite canreinvite=no nat=no qualify=yes port=5060 [102](internal-phones) username=102 secret=102 callerid=GNUbie102 [EMAIL PROTECTED] - - - s n i p - - - Thank you in advance. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro [EMAIL PROTECTED] wrote: Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Maybe I have my threads confused but I thought you got one way audio when three calls were made, you only mentioned one call. On Thu, Oct 16, 2008 at 12:44 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro [EMAIL PROTECTED] wrote: Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Sorry, wrong thread, time for bed. I thought this was the thread where the guy was having issues with one way audio on his third call, and his Asterisk server was behind NAT. Good night everyone and have pleasant dreams of 700 point drops in the DOW! OT, did you know if the government took the $700+ billion dollars and did not bail out the greedy banks, we could have immediate relief since for the most part, we could suspend Federal Income tax for everyone. A $300 rebate check, give me a break, how about some real stimulus, a rebate (or lack of theft because there is no law that we as individuals have to pay Federal Income tax, and I dare anyone to point it out, a real law, not something the IRS made up, I don't think they are part of the Legislative branch) weekly or bi-weekly depending on how you get paid. It would be immediate and give more money to the people who need it. All your Fed Income tax pays for anyways is the national debt, the clock just maxed out at $10 trillion. Rather than paying it down below the max and keeping it that way, they are building another one with additional digits. Sorry for a TOTALLY OFF topic post. I screwed up so I thought I might as well rant a little. Apologies in sheer exhaustion, Steve Totaro Thanks, Steve Totaro On Thu, Oct 16, 2008 at 12:46 AM, Steve Totaro [EMAIL PROTECTED] wrote: Maybe I have my threads confused but I thought you got one way audio when three calls were made, you only mentioned one call. On Thu, Oct 16, 2008 at 12:44 AM, GNUbie [EMAIL PROTECTED] wrote: Hello Steve, On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro [EMAIL PROTECTED] wrote: Did you try it the magic number of times, three? I'm sorry. What do you mean? Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Hello Steve, On Thu, Oct 16, 2008 at 12:42 PM, Steve Totaro [EMAIL PROTECTED] wrote: canreinvite defaults to yes, whether specified or not. http://www.voip-info.org/wiki/view/tips If you follow these directions adapting to your particular circumstances and it doesn't work, post your whole sip.conf Start asterisk with verbose set to 3 or so and turn on sip debugging. I get somewhere in the debug, you will see local NAT IPs that don't belong there, or it will just work. My /etc/asterisk/sip.conf is at http://lists.digium.com/pipermail/asterisk-users/2008-October/220256.html and my SIP phone is located within the LAN where the Asterisk box is also part of it. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
