Re: [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)
On Sat, Jun 19, 2010 at 12:00 PM, James Lamanna jlama...@gmail.com wrote: On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think this is a bug: https://issues.asterisk.org/view.php?id=17532 Has anyone dealt with this at all? Thanks. -- James Hello james, in the SPA config webpage on EXT 1 in the Nat Keep Alive MSG you should set $OPTIONS instead of $NOTIFY. then in your asterisk extension default context just set this: exten = s,1,Hangup then the phone will send a options packet and you will get a 200 OK instead of 489 Bad event. this should help. best regards Thanks Steve, I'll give that a try. I think I'll also look into why responses to NOTIFYs don't do the right thing in terms of NAT either. steve -- James I have created an issue report on this a few weeks on with Asterisk 1.6.2.8-rc1. This was happening on a client site, which I didn't have a chance to stop back by, so they closed the issue. https://issues.asterisk.org/bug_view_page.php?bug_id=17379 It looked to me like Asterisk was rejecting the NOTIFY message due to no callid, which is in the message. I couldn't figure out what was going and there is code in 1.6.2.x to return a 200 OK to a NOTIFY message. Ryan -- _ -- 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] applicationmap and ChannelRedirect
Hi, Does the lack of answers prove that the problem described in bug 17117 isn't a problem in reality and everything is caused by my setup? /PH -- Per-Henrik Lundblom epost: p...@whatever.nu telefon: 0733-20 71 26hemsida: www.whatever.nu -- _ -- 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] applicationmap and ChannelRedirect
Hi, It is hard to just look at your code and figure out why it is not working, and personally I don't have time to test it on an asterisk system and suggest you something. But I can tell you that I have implemented a couple of times somewhat similar dialplan functionality, recent of which was about 6 months ago. I used the n-way calling example as a reference and first made a working n-way feature, which itself was tricky, and then modified it for fit my requirement, which again, took quite a bit of debugging to make it work. So yes, I think it is somewhere in your dial plan that you should look. Have you tried to implement the n-way calling feature? That helped me a lot to understand how to successfully bridge in-progress calls to other channels, and make dynamic feature codes to work with it. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-20 9:31 AM, Per-Henrik Lundblom p...@whatever.nu wrote: Hi, Does the lack of answers prove that the problem described in bug 17117 isn't a problem in reality and everything is caused by my setup? /PH -- Per-Henrik Lundblom epost: p...@whatever.nu telefon: 0733-20 71 26 hemsida: ... -- _ -- 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] load balance meetme
Hello wise Group, if i plan to load balance incoming SIP Calls for MeetMe Conference to 2 or more Server, i think it is a Problem, because each Server opened his own MeetMe Room/Channel. Is it possible to made some interconnect the dahdi or MeetMe Channels over many Servers? (Like PHP, it can store the SessionID in a distributed filesystem or mysql database) Or what is the best way, to load balance Meetme conferences? I read abut openSIPS, thats sounds nice, but additionally i would like a kind of asterisk channel interconnect for better load balance. Thanx for your help. Daniel -- _ -- 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] Deleting some of the CDR data - How to do it safely?
Hi Guys, I am looking to delete some of the CDR logged by Asterisk in asteriskcdrdb in a PbxinaFlash system running Asterisk 1.4.x The CDR records to deleted are probably a big chunk and spread out all through the database but I basically want to delete all calls that came in through a specific DID. I think they all show as SIP/did_number I don't want to break the system or break the reporting tool of FreePBX so some advice would be appreciated. Also, any MySQL commands related to deleting such records with conditions above mentioned would be helpful. Thanks -- _ -- 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] Voicemail ODBC
On Saturday 19 June 2010 08:35:47 Zeeshan Zakaria wrote: On 2010-06-19 9:10 AM, Andraž atle...@gmail.com wrote: Ok, this issue I resolved, I just changed the TDS version to 7.0. But now I receive different error, I can't insert into database. [Jun 19 14:30:25] WARNING[6212] app_voicemail.c: SQL Prepare failed![INSERT INTO pbx_VoiceMail (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailb oxuser,mailboxcontext) VALUES (?,?, ? , ?,?,?,?,?,?,?)] [Jun 19 14:30:25] WARNING[6212] res_odbc.c: SQL Prepare failed. Attempting a reconnect... [Jun 19 14:30:25] NOTICE[6212] res_odbc.c: Connecting sqlserver [Jun 19 14:30:25] NOTICE[6212] res_odbc.c: res_odbc: Connected to sqlserver [kupalaodbc] Seems to me that the SQL statement is not complete and asterisk is complaining about it. Do you prepare this statement in your dialplan or asterisk makes it automatically? What, specifically, do you think is not complete about this prepare statement? The only error that I can guess at is that the table pbx_VoiceMail does not have all of the columns listed in the statement. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] dahdi span
On Saturday 19 June 2010 10:47:07 Daniel Knoll wrote: Hello Group, what does the Compiler Option mean LOTS_OF_SPANS ? The description is: More than 32 DAHDI spans Does this mean, more than 32 DAHDI Channels ? No, it means spans. If you only have analog DAHDI channels, then you should ignore this option. It is meant for those with PRI or SS7 links. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)
On Sun, Jun 20, 2010 at 5:42 AM, Ryan Wagoner rswago...@gmail.com wrote: On Sat, Jun 19, 2010 at 12:00 PM, James Lamanna jlama...@gmail.com wrote: On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think this is a bug: https://issues.asterisk.org/view.php?id=17532 Has anyone dealt with this at all? Thanks. -- James Hello james, in the SPA config webpage on EXT 1 in the Nat Keep Alive MSG you should set $OPTIONS instead of $NOTIFY. then in your asterisk extension default context just set this: exten = s,1,Hangup then the phone will send a options packet and you will get a 200 OK instead of 489 Bad event. this should help. best regards Thanks Steve, I'll give that a try. I think I'll also look into why responses to NOTIFYs don't do the right thing in terms of NAT either. steve -- James I have created an issue report on this a few weeks on with Asterisk 1.6.2.8-rc1. This was happening on a client site, which I didn't have a chance to stop back by, so they closed the issue. https://issues.asterisk.org/bug_view_page.php?bug_id=17379 It looked to me like Asterisk was rejecting the NOTIFY message due to no callid, which is in the message. I couldn't figure out what was going and there is code in 1.6.2.x to return a 200 OK to a NOTIFY message. Ryan Interesting. I'm still on the 1.4.x series (and I don't plan on upgrading until 1.8.x is out), but my issue, without the workaround that Steve suggested above, is that the NOTIFY Bad Event reply does not seem to respect NAT for some reason. Whether it doesn't look up the peer properties or what I'm not sure, but I plan on doing a thorough investigation with 1.4.32 this week to see what is indeed going on. Problems like this, and some other issues I've reported (where a channel can get stuck Up if a phone goes Unavailable while in Ringing), makes me lean more and more to moving to OpenSIPs for handling device registrations. -- James -- _ -- 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] Voicemail ODBC
If I use MySQL with the same fields it's working fine. I think that is something wrong with FreeTDS drivers. On Sun, Jun 20, 2010 at 7:13 PM, Tilghman Lesher tles...@digium.com wrote: On Saturday 19 June 2010 08:35:47 Zeeshan Zakaria wrote: On 2010-06-19 9:10 AM, Andraž atle...@gmail.com wrote: Ok, this issue I resolved, I just changed the TDS version to 7.0. But now I receive different error, I can't insert into database. [Jun 19 14:30:25] WARNING[6212] app_voicemail.c: SQL Prepare failed![INSERT INTO pbx_VoiceMail (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailb oxuser,mailboxcontext) VALUES (?,?, ? , ?,?,?,?,?,?,?)] [Jun 19 14:30:25] WARNING[6212] res_odbc.c: SQL Prepare failed. Attempting a reconnect... [Jun 19 14:30:25] NOTICE[6212] res_odbc.c: Connecting sqlserver [Jun 19 14:30:25] NOTICE[6212] res_odbc.c: res_odbc: Connected to sqlserver [kupalaodbc] Seems to me that the SQL statement is not complete and asterisk is complaining about it. Do you prepare this statement in your dialplan or asterisk makes it automatically? What, specifically, do you think is not complete about this prepare statement? The only error that I can guess at is that the table pbx_VoiceMail does not have all of the columns listed in the statement. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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
Re: [asterisk-users] Voicemail ODBC
On Sunday 20 June 2010 13:15:11 Andraž wrote: If I use MySQL with the same fields it's working fine. I think that is something wrong with FreeTDS drivers. Could also be that you're specifying the database name incorrectly. Are you able to see the tables when using the 'isql' command line tool? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Muti Asterisk
Hello, The CPU load is great...It's 90% idle...even memory is great...60% Idle Regards On Sat, Jun 19, 2010 at 7:36 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Jun 19, 2010 at 5:21 AM, michel freiha mich...@gmail.com wrote: Waiting your reply Reply: Do not cross-post to #asterisk-dev -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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
Re: [asterisk-users] dahdi span
ok, thanx for your answer. Daniel Am 20.06.2010 um 19:17 schrieb Tilghman Lesher: On Saturday 19 June 2010 10:47:07 Daniel Knoll wrote: Hello Group, what does the Compiler Option mean LOTS_OF_SPANS ? The description is: More than 32 DAHDI spans Does this mean, more than 32 DAHDI Channels ? No, it means spans. If you only have analog DAHDI channels, then you should ignore this option. It is meant for those with PRI or SS7 links. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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 Daniel Knoll Liberdastr. 9 12047 Berlin fon +49 (0)179 20 16 50 8 mail dan...@danielknoll.de web www.danielknoll.de -- _ -- 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] applicationmap and ChannelRedirect
* Zeeshan Zakaria zisha...@gmail.com [100620 16:14]: But I can tell you that I have implemented a couple of times somewhat similar dialplan functionality, recent of which was about 6 months ago. I used the n-way calling example as a reference and first made a working n-way feature, which itself was tricky, and then modified it for fit my requirement, which again, took quite a bit of debugging to make it work. So yes, I think it is somewhere in your dial plan that you should look. Have you tried to implement the n-way calling feature? That helped me a lot to understand how to successfully bridge in-progress calls to other channels, and make dynamic feature codes to work with it. Thanks for your answer. I haven't tried the n-way example, just looked at the dialplan and I think I have understood how it works. Will try the n-way example to see if I have missed something. Still the bug 17117 rings in the back of my head because it matches my problem... /PH -- Per-Henrik Lundblom epost: p...@whatever.nu telefon: 0733-20 71 26hemsida: www.whatever.nu -- _ -- 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] Corba interface
On 6/15/2010 9:57 AM, Muro, Sam wrote: Does anyone know how to configure asterisk to be able to query Corba interface directly from the dialplan I do it by implementing my dialplan as a Java server using the asterisk Java FastAGI interface, then from Java I can make all the CORBA calls I want (I personally use JacORB). Check http://www.lumenvox.com/partners/digium/applicationzone/projects/javaPizza.aspx for some sample code on the Java part - this happens to drive Lumenvox to handle a call, but you can easily hack it to stick CORBA calls in. Steve -- _ -- 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] Compiling H323
I'm really struggling with an Asterisk 1.6.2.7 install (on centos 5.4) The pwlib + opal packages don't satisfy Asterisk's configure script (to let H323 compile), so I removed those and added the latest ptlib + h323plus (from h323plus.org) I can compile ptlib and h323, but when I load chan_h323 in asterisk I get a segfault. I had to point LD_LIBRARY_PATH to /usr/local/lib with the lib files from ptlib and h323+ are installed by default, to even get asterisk to start. Can anyone give some advice? I've burned 8 hours on h323 today -- _ -- 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] Compiling H323
And after cleaning everything 323/pwlib/ptlib related, and reinstalling ptlib and h323plus, I can't even get asterisk to compile chan_h323 anymore. Perhaps something old was left over. My .configure run shows: checking /usr/src/openh323plus/h323plus/../pwlib/version.h usability... no checking /usr/src/openh323plus/h323plus/../pwlib/version.h presence... no checking for /usr/src/openh323plus/h323plus/../pwlib/version.h... no checking /root/pwlib/include/ptlib.h usability... yes checking /root/pwlib/include/ptlib.h presence... yes checking for /root/pwlib/include/ptlib.h... yes checking if PWLib version 2.4.5 is compatible with chan_h323... yes and checking h323.h usability... no checking h323.h presence... no checking for h323.h... no Anyone got good advice? I can make ooh323 work but a bug with faststart (in h323) is forcing me to another h323 stack. Is there a way to make opal+pwlib from centos packages work? (trick asterisk .configure to accept them)? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis [mdup...@ocg.ca] Sent: Sunday, June 20, 2010 7:45 PM To: Asterisk Users List Subject: [asterisk-users] Compiling H323 I'm really struggling with an Asterisk 1.6.2.7 install (on centos 5.4) The pwlib + opal packages don't satisfy Asterisk's configure script (to let H323 compile), so I removed those and added the latest ptlib + h323plus (from h323plus.org) I can compile ptlib and h323, but when I load chan_h323 in asterisk I get a segfault. I had to point LD_LIBRARY_PATH to /usr/local/lib with the lib files from ptlib and h323+ are installed by default, to even get asterisk to start. Can anyone give some advice? I've burned 8 hours on h323 today -- _ -- 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] Asterisk 1.6 + Jabber crashes
Hello, I am attempting to setup Asterisk to work with Gtalk. I am using the following versions: Slackware Linux 12.0 Asterisk 1.6.2.9 GNU TLS 2.8.6 Iksemel (svn v25) OpenSSL 0.9.8o It all compiles however about 10 seconds after starting Asterisk it crashes. To mitigate this issue I have moved jabber.conf to another directory and then Asterisk starts up. So I assume the issue is with this file? I can't find any issues: jabber.conf: [general] debug=no autoprune=no autoregister=no [gtalk1] type=client serverhost=talk.google.com username=my_gm...@gmail.com/Talk secret=MY_PASS port=5222 usetls=yes usesasl=yes ;buddy=my_fri...@gmail.com ;statusmessage=This is an Asterisk server ;timeout=100 gtalk.conf: [general] context=default allowguest=no bindaddr=IPADDR [my-Friend-context] username=my_fri...@gmail.com disallow=all allow=ulaw context=custom-michael connection=gtalk1 When jabber.conf is removed from /etc/asterisk the server starts up and runs properly. I have spent a lot of time Googling this issue though most information seems to be about Asterisk 1.4, so I would have expected any issues there to have been fixed long ago? If there is any other information needed please advise. Michael -- _ -- 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
