[Asterisk-Users] Open files / socket leak
Title: Open files / socket leak We're using STABLE CVS-Nv1-0-5-02/24/05 and we've been noticing that sometimes there's a socket leak on REGISTER SIP messages. We've seen it happen only on customers using Sipura SPA2100 ATAs. If I issue a sip show channels, I see thousands of zombie channels. If I look into the details, that's what I get - actually one single sip show channel channelID returns thousands of these: * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED]: 520 REGISTER Our Codec Capability: 12 Non-Codec Capability: 1 Their Codec Capability: 0 Joint Codec Capability: 0 Format unknown Theoretical Address: x.x.x.x:5060 Received Address: x.x.x.x:5060 NAT Support: RFC3581 Our Tag: 715659627 Their Tag: SIP User agent: Need Destroy: 0 Last Message: Promiscuous Redir: No Route: N/A DTMF Mode: rfc2833 The sequence number (ie. 520) increases by 1 every time. After a while, I run out of files and I have to restart asterisk. I have temporarily solved the problem by issuing a ulimit -n 8192 in safe_asterisk, but that's not a solution, since I will eventually reach that limit as well. Is there a way to fix this? We're running RHEL4 and we have about 300 customers registered all the time. Thank you very much -Manuel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_odbc logging insane integer values
Title: cdr_odbc logging insane integer values
I'm having a problem with * (tried both HEAD and STABLE). When logging with cdr_odbc through unixODBC to MySQL, I get insane integer values in the duration, billsec, disposition and amaflags fields. I have enabled MySQL logging, and that's the query I get:
INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield) VALUES ('2005-02-23 11:47:04','1234','1234','161','auth-out','SIP/xxx','','Hangup','',8589934594,17179869186,12884901892,3630868242827837443,'xxx','1109155624.0','161')
I've tried MySQL 4.1.7 or 4.1.10, with unixODBC 2.2.9 or 2.2.10, with MyODBC 3.51.11 or 2.50.39. All of these give the same error.
Logging to csv file works correctly. I have also tried inserting a debug line in cdr_odbc.c, to print out the values of cdr-duration etc, and they are all correct. The problem seems to happen somewhere between * and MySQL.
The only thing I have on this system which is different than all my other systems is the platform: x86_64 (AMD Opteron) instead of 32bit i686.
Anyone?
Thank you
-Manuel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segfault when using res_config_odbc on x86_64
Title: Segfault when using res_config_odbc on x86_64 I'm trying to move our asterisk setup from an i686 server to an x86_64 (Dual AMD Opteron) server. Everything has been manually compiled: MySQL 4.1.10, MyODBC 3.51.11, unixODBC 2.2.10 (because I couldn't find any usable RPMs). And obviously Asterisk, where I'm using the STABLE release CVS-Nv1-0-5-02/21/05. The OS is Redhat Enterprise ES 4.0, kernel 2.6.9-5.0.3.ELsmp. Asterisk is configured to read sip.conf and extensions.conf from MySQL via ODBC. The ast_config table contains about 3500 rows. When I start asterisk -gcv the first time, everything works fine. Then I issue a reload, and something strange happens: Reloading SIP Feb 22 09:50:23 NOTICE[26316]: config.c:764 __ast_load: Loading Config sip.conf via odbc engine == Parsing '/etc/asterisk/res_config_odbc.conf': Found Use EXIT or QUIT to exit the asterisk console *CLI After that, basically nothing else happens, and asterisk still works. Now, if I issue another reload, asterisk segfaults. That's what bt says inside gdb: (gdb) bt #0 0x002a9607df81 in my_strcasecmp_8bit () from /usr/local/lib/mysql/libmysqlclient_r.so.14 #1 0x002a96075e71 in get_charset_number () from /usr/local/lib/mysql/libmysqlclient_r.so.14 #2 0x002a960761a6 in get_charset_by_csname () from /usr/local/lib/mysql/libmysqlclient_r.so.14 #3 0x002a960909a2 in mysql_real_connect () from /usr/local/lib/mysql/libmysqlclient_r.so.14 #4 0x002a95f28ba3 in SQLConnect () from /usr/local/lib/libmyodbc3_r.so #5 0x002a95baf4e9 in SQLConnect (connection_handle=0x655380, server_name=0x6007f0 asterisk, name_length1=-4992, user_name=0x770300 asterisk, name_length2=-6000, authentication=0x60cfb0 ***, name_length3=-3) at SQLConnect.c:3796 #6 0x002a9bd3e50b in odbc_init () at cdr_odbc.c:383 #7 0x002a9bd3e877 in odbc_load_module () at cdr_odbc.c:305 #8 0x00411204 in ast_module_reload (name=0x0) at loader.c:184 #9 0x0042f811 in handle_reload (fd=-1776674848, argc=1702130529, argv=0x6146a0) at cli.c:127 #10 0x00431b48 in ast_cli_command (fd=1, s=0x2a961a17e0 \b) at cli.c:1155 #11 0x00447334 in main (argc=5986672, argv=0x5ae0d0) at asterisk.c:706 I'm not sure what the problem could be. What's interesting is that I can issue sip reload or extensions reload as much as I want and it won't segfault. But two plain reloads in a row will crash it. Anyone? Thank you -Manuel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Solaris 10
Title: Asterisk on Solaris 10 Does anyone have experience compiling Asterisk STABLE 1.0.5 on Solaris 10 for x86? I have looked at http://www.voip-info.org/wiki-Asterisk+Solaris+Support but I'm looking for other people's experience in actually using Asterisk under that platform. We only need SIP and IAX2 channels. Is the STABLE even supposed to compile? The above mentioned page says Since 15/Dec/2004, the CVS HEAD version of asterisk has included support for Solaris. Solaris support is not yet included in the stable releases. What compiler am I supposed to use? Do I need to install gcc? Any help or comments are appreciated. Thank you -Manuel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HEAD vs STABLE
Title: HEAD vs STABLE I am having a somewhat hard time finding out what the current HEAD and STABLE versions of Asterisk are. I'm currently running CVS-HEAD-08/24/04 and I think I need something newer :-) What are the version numbers of HEAD and STABLE? I was unable to find out by reading here and there. I've seen there's a 1.0 stable, a 1.0.5 (is this head or stable?), and I'm not sure what to download in order to have something stable *and* not too old. Sorry if this might seem a dumb question, but since I haven't been updating for a few months now, I might have missed something :) Thank you -Manuel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AS5xx0: SS7 and SIP?
Title: AS5xx0: SS7 and SIP? We currently use Asterisk to provide a SIP-to-PSTN service. The actual conversion takes place somewhere in a softswitch owned by our SIP-to-PSTN provider, where we have an SS7 link. We would like to do that conversion ourselves. Is it possible to replace a softswitch with a Cisco AS5xx0 only (ie. AS5300, 5350, 5400), or is a *real* softswitch (ie. Cisco PGW2200) needed? Does anyone have any experience with an Asterisk--CiscoAS5xx0--SS7 configuration? As far as I know, Asterisk--CiscoAS5xx0--PRI works, but I couldn't find anything about SS7. Thank you -Manuel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple IPs and SIP
Title: Multiple IPs and SIP This topic has already been brought up lately, but I'd like to inquire if there are any news. I have 2 IP addresses assigned to my ethernet card (eth0 and eth0:0) that are on 2 different subnets (one public, one private), because our PSTN gateway provider has established a private link with us using private addresses. Yes, that's not exactly clean, 2 ethernet cards would be better, but it shouldn't make a difference for this issue. Now the problem: I have asterisk bound to 0.0.0.0 in sip.conf, so it binds to every address I have. The routes in my system are configured correctly. But as soon as asterisk tries to send a SIP packet to a private address, using eth0:0, the following happens: - the packet actually arrives at my PSTN provider's border gateway - the sender IP address in the UDP packet is correct, it's my private address, so linux did its routing job correctly - the SIP packet contains my public (!) address in the From and Contact fields, which is wrong, and confuses our PSTN provider's gateway, which in turn tries to send the reply to our public address, which it will never find, because their server is on a private network only This has to do with the fact that SIP wants to encode ethernet packet information in a higher layer, which is somewhat stupid IMHO, but I think we have to live with this :) We're using CVS-HEAD-08/24/04 (yes, I have to upgrade sometimes...). Has this issue been addressed lately, or is it somehow unsolvable? Thank you everyone -Manuel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] problem with zyxel prestige 2002
Title: R: [Asterisk-Users] problem with zyxel prestige 2002 We also had this problem with the zyxel 2002. Upgrade to the latest firmware, then it will work. Older firmwares had trouble with incoming calls behind NAT. -Manuel -Messaggio originale- Da: Stig Thune [mailto:[EMAIL PROTECTED]] Inviato: lunedì, 15. novembre 2004 19:16 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] problem with zyxel prestige 2002 This sounds odd. We use the same adapter. I will check this more.. Are u sure you have set the phone up correctly ? And also - have to checked the ring phone1 or phone2 on incomming calls ? / Stig Henning - Original Message - From: Mihkel Raba [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 9:51 PM Subject: [Asterisk-Users] problem with zyxel prestige 2002 Hi I tried to use Zyxel Prestige 2002 VoIP Analog Telephone Adapter with asterisk. Device registers both phones and i can call out. But incoming calls are not working. Asterisk - sip show peers shows zyxel, zyxel web interfce shows that devices are registered. But when i do incoming call to zyxel, phones do not ring and if voicemail is configured, calls go directly to voicemail. Any suggestions ? Mihkel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zyxel Prestige 2002/2002L sound quality
Title: Zyxel Prestige 2002/2002L sound quality Hi everyone, I've been trying a Zyxel Prestige 2002L ATA with Asterisk, but I have a problem with very bad sound quality (using G711, it sounds very robotic and metallic) and there is a very long delay in the audio. This all doesn't happen with other ATAs (like the Grandstream HT286). Apparently there are no parameters you can set. I already have the latest firmware installed. Could it be a defective unit? Does anyone have experience with this ATA? Thanks -Manuel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hung SIP channels
I have recently posted a message regarding hung SIP channels when using the Monitor() command. Well, I was wrong. The channel hanging wasn't caused by the Monitor command. They also hang when they aren't monitored. The cause seems to be our PSTN gateway provider. When for some reason their PSTN gateway crashes or reboots (OK, this should happen, but anyway...) and RTP/SIP data stops flowing, the SIP channel just sits there on the Asterisk server, and will never hang up, until you soft hangup it. When issuing show channel SIP/xxx on a hung channel, you see the elapsed seconds counter incrementing, but the frames in and frames out counters don't increment anymore. When you soft hangup the channel, the macro it was running in succesfully continues to the hangup extension and everything is cleaned up properly. Is there a way to tell Asterisk to hang up stuck channels automatically, ie. when no frames are received for more than, say, 30 seconds? We're using CVS-HEAD-08/24/04 Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Grandstream Firmware
What is the current stable firmware version? 1.0.5.11 Do the ATA's and the phones use the same firmware? Yes, it's the same firmware -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitor() hangs
We use the Monitor() command to record all incoming calls to our call center. After about 100 incoming calls, the Monitor() command starts to hang, as follows: - a call comes in - Asterisk starts recording, the -in.wav and -out.wav files are created - the partys talk - after a while (between 1 and 15 minutes), both partys hear only silence (as in no RTP data flows). - when this happens, Asterisk stops recording (the wav files don't grow anymore, they just sit there open). - when this happens, all other SIP calls active at that time suffer the same problem (ie. the callers don't hear each other anymore) - the Monitor()ed call's channels remain there, as if the call were still active, even after the caller hangs up (Asterisk ignores the hangup) - if I soft hangup the channels, Asterisk finally destroys the channels, writes the CDR record, closes the wav files and soxmix'es them. - the other, non Monitor()ed SIP calls that suffered the silence problem are hung up correctly if one of the 2 partys hangs up (ie. the channels don't hang there forever like the Monitor()ed call) - if I listen to the recorded wav file, the recording just stops abruptly at the end, no click, no anything - not even silence. It just stops I'm quite clueless. A July CVS-HEAD is exhibiting the same behaviour. We are using SIP only, nothing fancy. The PSTN gateway is somewhere else, not on Asterisk. We are using CVS-HEAD-08/24/04-08:09:55 on Fedora Core 1 (Kernel 2.4.22-1.2188.nptlsmp) on a Dual Xeon server (HT disabled). We never have more than about 20 channels active at a time, so I don't think the system is too loaded. Anyone? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT table expiration
I'm having a problem with some customers sitting behind hopefully SIP aware routers doing NAT. These routers translate port 5060 to something different (ie. 10001) in order to be able to connect more than one SIP client on a single NATted LAN. Unfortunately, after a while the router seems to forget the NAT table, ie. which port belongs to which SIP client on the LAN. This happens with different router brands (Alcatel, Zyxel). Maybe the expiration is set too long? Is there a common standard or a maximum recommended value for these situations? I think 60 minutes is definitely too much, 5 minutes is too short... is there a better way than trial and error to find out a good value for this? Another question: in a sip show peer screen i see the following values Expire : 27345 Expiry : 900 What's the difference between them? Furthermore: if the client supports STUN or other NAT traversal methods, do I need to use them, or will Asterisk take care of everything? We always have Asterisk in the media path, so clients never talk to each other directly. Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Dial plan errors
I'm having the same problem here. Any solution to this problem?
-Manuel
(sorry for top-posting, I'm having a stupid mail client here)
-Messaggio originale-
Da: Simon Brown [mailto:[EMAIL PROTECTED]
Inviato: giovedì, 1. luglio 2004 02:05
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users] Dial plan errors
I am attempting to implement the new features added recently where you can have
Goto(s-DIALSTATUS) in the dial plan. My extensions.conf looks like this: exten =
s,1,Dial(${ARG2},20,r)
exten = s,2,Goto(s-${DIALSTATUS})
exten = s-NOANSWER,1,Voicemail(u${ARG1})
exten = s-CHANUNAVAIL,1,Voicemail(b${ARG1})
exten = s-BUSY,1,Voicemail(b${ARG1})
exten = s-.,1,Voicemail(u${ARG1})
But this is what happens when no-one answers:
-- Executing Macro(SIP/201-c410, stdexten|201|SIP/201) in new stack
-- Executing Dial(SIP/201-c410, SIP/201|20|r) in new stack
-- Called 201
-- SIP/201-4d1f is ringing
-- Nobody picked up in 2 ms
-- Executing Goto(SIP/201-c410, s-NOANSWER) in new stack
-- Goto (macro-stdexten,s,0)
-- Timeout on SIP/201-c410
== CDR updated on SIP/201-c410
-- Executing Hangup(SIP/201-c410, ) in new stack
== Spawn extension (sip, t, 1) exited non-zero on 'SIP/201-c410'
It tries to Goto (macro-stdexten,s,0)
___
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to make * don't strip the leading 0
Is it possible to tell asterisk not to strip the leading 0
of *incoming* MSNs? I use asterisk with i4l and whenever
I get a call from an long-distance party, the leading 0, which
should be there according the german numbering, is not.
Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped
already by the phone company? I think the easiest thing for you would be to add the
leading 0 before forwarding the call to your SIP client (ie.
SetCallerID(0${CALLERIDNUM}) in your extensions.conf for each extesion where you'd
like to add the 0).
Regards
Manuel
___
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] VoIP hackers gut Caller ID
hi... here in Italy is almost impossible to set an invalid cid, if is out of your allowed space. ie. if you have X numbers on your PRI, you can only set one of these. nothing more. on bri you simply cannot do nothing. just my 2 cents. In Switzerland CLI is also impossible to spoof - by default. If you ask the BRI/PRI provider, and you have an ISDN connection with DDI, they enable CLIP Special Arrangement, which allows to add a presentation number to the real CLI. So you can't really abuse of it, because your real number is always transmitted together with your pretend-to-be CLI. The advantage of this is that anyone can change his CLI, for example to make outgoing calls and show a 0800 number in the customer's cell phone. We use this feature in our company, because our customers know us by our 0800 number, not the real number hiding behind it. The disadvantage is that not all networks accept presentation numbers, for example Orange Mobile. In this case, the caller's real CLI will be displayed instead of the presentation number. If you get yourself an SS7 link that's a different story, but in this case you're supposed to be a trusted entity, and you shall not spoof and play with numbers that you're not allowed to use. IMHO, trusted entities with SS7 links that abuse of that power should simply be disconnected from the public network. Not every kid with a couple $1000 spare should be allowed to play with this. -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Software SIP fax client
Does anyone know of a software SIP fax client? Something I can install on a PC which connects to the asterisk server and sends/receives faxes? Something like XLite - but to fax instead of to phone. I know of the fax machine connected to an ATA solution, but that's not really what I'm looking for :-) Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_odbc not working
I have been playing with res_odbc in these last days, but it doesn't want to work. This is the output when starting Asterisk, so everything seems OK: [res_odbc.so] = (ODBC Resource) == Parsing '/etc/asterisk/res_odbc.conf': Found Jul 7 20:11:32 NOTICE[-1084915040]: res_odbc.c:132 load_odbc_config: registered database handle 'mysql' dsn-[MySQL-asterisk] Jul 7 20:11:32 NOTICE[-1084915040]: res_odbc.c:369 odbc_obj_connect: res_odbc: Connected to mysql [MySQL-asterisk] Jul 7 20:11:32 NOTICE[-1084915040]: res_odbc.c:402 load_module: res_odbc loaded. [res_config_odbc.so] = (Odbc Configuration) Jul 7 20:11:32 NOTICE[-1084915040]: config.c:880 ast_config_register: Registered Config Engine odbc == Parsing '/etc/asterisk/extconfig.conf': Found Jul 7 20:11:32 NOTICE[-1084915040]: res_config_odbc.c:233 load_module: res_config_odbc loaded. These are the pertintent config files: ; extconfig.conf [settings] queues.conf = config_odbc sip.conf = config_odbc ; res_odbc.conf [mysql] dsn=MySQL-asterisk username=*** password=*** pre-connect=yes ; [settings] table = ast_config connection = mysql And that's what I have in my ast_config table: ++++---+-+---+-+-+ | id | cat_metric | var_metric | commented | filename| category | var_name| var_val| ++++---+-+---+-+-+ | 27 | 0 | 0 | 0 | queues.conf | test | Member | Agent/1000,1234,1 | 1 | 0 | 0 | 0 | sip.conf| general | port| 5060 | 2 | 0 | 0 | 0 | sip.conf| general | bindaddr| 0.0.0.0 | 3 | 0 | 0 | 0 | sip.conf| general | context | default | 6 | 0 | 0 | 0 | sip.conf| general | disallow| all | 7 | 0 | 0 | 0 | sip.conf| general | allow | ulaw | 8 | 0 | 0 | 0 | sip.conf| general | allow | alaw | 9 | 0 | 0 | 0 | sip.conf| general | allow | g729 | 19 | 0 | 0 | 0 | sip.conf| 20190 | type| friend | 20 | 0 | 0 | 0 | sip.conf| 20190 | username| 20190 | 21 | 0 | 0 | 0 | sip.conf| 20190 | secret | secret | 22 | 0 | 0 | 0 | sip.conf| 20190 | host| dynamic | 23 | 0 | 0 | 0 | sip.conf| 20190 | context | auth-out | 24 | 0 | 0 | 0 | sip.conf| 20190 | canreinvite | no | 25 | 0 | 0 | 0 | sip.conf| 20190 | callerid| 1234567 | 26 | 0 | 0 | 0 | sip.conf| 20190 | accountcode | 20190 The problem is that all of this is ignored. Show queues is empty, and no peers are added, I don't see anything in sip show peers. What did I forget? :-) I don't think it's an ODBC problem, because I use the same datasource for CDR logging. Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ringinbacktone even without 'r', and inexistant codec
I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk. It works ... Partially. We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA. First of all, when I make a SIP call to the unit with a simple Dial() command (no r, so Asterisk shouldn't generate its ringback tone) I hear Asterisk's ringback tone anyway (I'm sure it's Asterisk generating it because by changing the country it indications.conf, the ringing changes). That's what I see in the CLI: -- Executing Dial(SIP/2017-71be, SIP/[EMAIL PROTECTED]|90) in new stack -- Called [EMAIL PROTECTED] -- SIP/inalp-eaf3 is making progress passing it to SIP/2017-71be -- SIP/inalp-eaf3 is ringing Now comes the fun part: if the ISDN extension answers the phone, the call is dropped, and I get the following message: -- SIP/inalp-eaf3 answered SIP/2017-71be -- Attempting native bridge of SIP/2017-71be and SIP/inalp-eaf3 -- Attempting native bridge of SIP/2017-71be and SIP/inalp-eaf3 Jul 7 20:36:20 WARNING[112708528]: chan_sip.c:1800 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 8/4) Now, frame type 64 is 16 bit signed linear PCM, which IMHO has nothing to do with SIP and its RTP stream. I have the usual disallow=all; allow=ulaw; allow=alaw sequence in sip.conf, and the Inalp unit is configured to allow alaw and ulaw, nothing else (it doesn't even support that 16 bit PCM thing). But we're not through yet. If I add the r paramenter to the Dial() command, the call completes successfully. But unfortunately, now Asterisk doesn't (!) generate the ringback tone anymore. I get no ringing at all, just silence, until the other party answers. Isn't * supposed to generate a ringback tone when r is appended in the Dial command? Isn't * supposed *not* to generate a ringback tone when there is *no* r? What in the world is codec 64? By the way, outgoing (ISDN-to-SIP) calls from the Inalp unit work perfectly. Other SIP clients (ATAs, softphones) work perfectly on our setup (and also the ringback tone behaviour is correct with those). Only that single unit (the most expensive one, by the way :-)) doesn't want to cooperate. I'm clueless here... Anyone? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] execute a context from cron
I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I would have a look at GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime That should be much easier than a cron job Regards -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Asterisk Docs
Timeout, but no rule 't' in context 'home'
about this line:
exten = 2201,1,Dial(${PHONES1},20,Ttm)
I know the problem is with the 't' but I don't know
what the parameters mean. I looking for a man page basically.
The problem isn't related to the t in the Dial() command, which enables call
transfer, but to a missing t (timeout) extension. More can be found here:
http://www.voip-info.org/wiki-Asterisk+standard+extensions
The voip-info.org site is a good reference if you're looking for something like a man
page for Asterisk.
-Manuel
___
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Which Linux ?
Based on th wiki, avoid kernel 2.6 unless you know what you are doing. Likewise with fedora, which seems to work but needs kernel thread turned off. Just my experience: I have installed Asterisk twice on Fedora Core 1 with kernel 2.4.22-1.2188.nptlsmp on Dual Xeon systems. It has worked perfectly both times, without needing any additional compiler flags, and no kernel panics. What I have found out is that I had to disable hyperthreading, or I would be getting very choppy audio (I think that's what you mean when you say needs kernel thread turned off). By the way, the noht flag in lilo/grub isn't enough, it has to be disabled in the BIOS. Don't know if that's an issue on other Linuxes as well. -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to force G729
We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway
provider supports both, so that's not a problem, and if I force him (the PSTN gateway)
to allow G729 only, the outgoing call will take place with G729.
The problem is that I want to have my PSTN provider configured to allow ULAW as a
first priority, then G729. I did it like that:
[mypstngate]
type=friend
host=192.168.0.100
port=5060
context=pstn-in
canreinvite=no
disallow=all
allow=ulaw
allow=g729
Then, in the outgoing context for our G729 SIP customers, I've put something like that:
exten = _0N,1,setvar(SIP_CODEC=g729)
exten = _0N,2,Dial(SIP/0041${EXTEN:[EMAIL PROTECTED],90)
What happens now when placing a call is very interesting. As you can see, Asterisk
wants to change the codec to g729, but on the outgoing call to the PSTN gateway it
remains ULAW. Like this, I'm using up one of my G729 licenses, and Asterisk is doing
the transcoding between G729 and ULAW. That's definitely not what I want. Any ideas
about how to force both channels to G729? By the way, if I use a client which doesn't
support G729, this call doesn't even take place, it hangs up, because Asterisk tries
to force G729 on the client's channel (but not on the PSTN gateway's channel).
In other words, the setvar(SIP_CODEC=g729) only forces the codec on the calling
channel, not on the called channel. How can I change that?
Another interesting thing, the show g729 after the call hangs up: I have -1/-2
encoders/decoders in use. Maybe a bug?
Thanks
-Manuel
*CLI -- Executing SetVar(SIP/2016-b119, SIP_CODEC=g729) in new stack
-- Executing Dial(SIP/2016-b119, SIP/[EMAIL PROTECTED]|90) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/mypstngate-caed is making progress passing it to SIP/2016-b119
-- SIP/mypstngate-caed is ringing
-- SIP/mypstngate-caed answered SIP/2016-b119
Jun 24 09:49:23 NOTICE[1094450096]: chan_sip.c:1314 sip_answer: Changing codec to
'g729' for this call because of ${SIP_CODEC) variable
-- Attempting native bridge of SIP/2016-b119 and SIP/mypstngate-caed
*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format
192.168.0.1000041911234 1f7d34e3642 00102/0 0ms ms ULAW
192.168.0.2 20164977-4F41-7 00101/3 0ms ms G729A
2 active SIP channel(s)
[... after hangup ...]
== Spawn extension (auth-out, 0911234567, 2) exited non-zero on 'SIP/2016-b119'
-- Executing Hangup(SIP/2016-b119, ) in new stack
== Spawn extension (auth-out, h, 1) exited non-zero on 'SIP/2016-b119'
cdr_odbc: Query Successful!
*CLI show g729
-1/-2 encoders/decoders of 30 licensed channels are currently in use
*CLI
___
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
Try to configure in sip.conf your extensions context like this: [XXX] disallow=all allow=g729 Done that already: but then, the incoming channel (from the user to Asterisk) is G729, and the outgoing channel (from Asterisk to the PSTN gateway) still remains ULAW, so Asterisk has to do transcoding (ULAW to G729), which I don't want, obviously. For some reason, the outgoing channel doesn't follow the setvar(SIP_CODEC=g729) rule. -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: [Asterisk-Users] How to force G729
Define that per user. Of course... The user part is not the problem. If I force a user in its extensions to use G729 only, he actually talks G729 to Asterisk, but asterisk still talks ULAW to the PSTN gateway, doing the transcoding. This is driving me crazy... -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anonymity and Privacy headers
When a user calling over the PSTN network calls one of our SIP users with a restricted number (CLIR), our PSTN gateway is sending us incoming calls with the following additional headers: Proxy-Require: privacy Anonymity: uri Remote-Party-ID: sip:[EMAIL PROTECTED]:4000;privacy=uri as opposed to Remote-Party-ID: sip:[EMAIL PROTECTED]:4000;privacy=off when CLIP is enabled (thus CLIR is disabled). Any ideas on how I can tell asterisk to process one (or more) of these headers, and strip the CLI before sending the call out to our SIP users, in case it is restricted? I have searched the Wiki and read the chan_sip.c source code, but didn't find anything useful... Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: R: [Asterisk-Users] How to force G729
If I understood your initial objective correctly (and I may not have), the user's phones are negotiating the codec to be used for each rtp session. Asterisk parameters can be used to dictate rtp sessions between the sip phone and asterisk, but that won't influence the next step in which the sip phone negotiates a new rtp session directly with the gateway. The gateway and the phone will negotiate a common codec based on whatever logic those two devices have been programmed with by their respective manufacturers; asterisk isn't involved. So, it sounds like the issue is understanding the codec selection logic that has been programmed into the gateway and the phone. I think you're getting my point, at least I think so (I'm getting more and more confused myself about this...) The problem is that the phone negotiates a codec with asterisk when placing the call (remember I have all reinvite's set to no, so the gateway and the phone won't talk directly to each other!). This negotiation actually works correctly, because I force the phone's codec using disallow=all; allow=g729 in the SIP phone's peer configuration. The negotiation which doesn't work the way I want is the asterisk-to-gateway part. The gateway can talk either ULAW or G729, whatever I tell it, if I force it using the disallow/allow method in sip.conf. The problem is that I need asterisk to talk to the gateway sometimes with ULAW, sometimes with G729, depending on the SIP phone who placed the call in the first place. What I need is some sort of command which says OK, now Dial(... @gateway), but force G729 (which works *if* I tell asterisk that the gateway supports G729 *only* in sip.conf, but we want it to support both codecs, right?). Apparently I can only force the codec on incoming channels, not on outgoing channels. Is this really an asterisk limitation? -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
Did you try having two sip.conf entries for your gateway? Forcing one with G729 and the other with ulaw? You would obviously need to change your dialplan accordingly and have each phone configured so that it would take the proper extension. I have not tried this, it is just really an idea... That's actually a very good idea, and I have tried it: for outgoing calls it works like charm. But then the problem is transferred to incoming calls (from the gateway-asterisk-SIP client). Because the gateway now has 2 entries, asterisk is confused about what codec it has to use for incoming calls, and for some reason I can't force it, because the 2 entries have the same IP. I'm starting to think that I won't be able to solve that myself, but that someone will have to program something for this to work... But if I'm the only one having this kind of request, I'm not too optimistic -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
Hmmm, I was thinking about this problem too... What type of gateway are you using? Is it registering with the Asterisk server? I would try using two different 'virtual' extensions on the gateway and in sip.conf. That way you would have full control on how calls from the gw to * are handled. I had thought about that, too ... Unfortunately the gateway is unable to register. We authenticate based on the IP address only. Otherwise, like you say, I could have 2 virtual extensions, but with IP only this is not possible. Maybe I will find a solution by sleeping over the problem (not physically, that is) tonight :-) -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
They don't need to have the same IP. Assign several IP numbers to your linux box: ifconfig eth0:1 10.1.1.1 netmask 255.255.255.0 ifconfig eth0:2 10.1.1.2 netmask 255.255.255.0 Sorry guys... These are all great tips, but also this doesn't work: the gateway is not under my control, it is actually a real phone switch, which isn't owned by us. Unfortunately I can't tell them to add a second IP ... :-) -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] How to force G729
Use two separate entries with type=peer and type=user instead of one
entry with type=friend.
Tried that as well. This triggers yet another misbehaviour...
I tried to define 2 peers (for the outgoing calls), one called [gateway-g729] and one
called [gateway-ulaw], each allowing only the codec specified in the name. Then I
defined 1 user for incoming calls from the gateway (called [gateway-in]), with both
g729 and ulaw in the allow list.
And you know what happens? Outgoing calls are now fine (I can direct them either to
@gateway-g729 or @gateway-ulaw in the Dial() command), but incoming calls seem to have
a live on their own, and choose whatever codec they prefer. Even if I
setvar(SIP_CODEC=ulaw), the gateway-to-asterisk channel seems to remain in g729 (at
least that's what I can tell from show g729 - because sip show channels looks
correct, both ULAW).
At some point I get that message:
Jun 24 16:37:14 NOTICE[1104739248]: chan_sip.c:1314 sip_answer: Changing codec to
'ulaw' for this call because of ${SIP_CODEC) variable
And yes, in sip show channels the gateway-to-asterisk channel is marked as ULAW, but
for some reason a G729 license is used up, because the call did start in G729... Any
ideas?
I guess I'm very close to the solution, but now G729 licenses are acting weird and are
being used even in ULAW-to-ULAW calls which started with G729 in the beginning...
-Manuel
___
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CSV log stopping
If I have ODBC logging enabled (with cdr_odbc), Asterisk logs everything to ODBC *and* to the CSV file (Master.csv). If I issue a reload, it stops logging to the CSV file, but continues logging to ODBC. To have it log to the CSV file again, I have to issue unload cdr_csv.so then load cdr_csv.so. Is that normal behaviour? Is it supposed to log to the CSV file with ODBC enabled or not? I'm using CVS-06/21/04-22:36:21 Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Accountcode missing in log
I have defined a SIP friend without username and secret, only IP-based. I have also defined an accountcode for that friend, as follows: [mypeer] type=friend host=192.168.0.100 port=5060 context=mycontext canreinvite=no accountcode=mypeer Unfortunately the accountcode for the calls originating from mypeer doesn't show up in the log (either CSV or ODBC). All the other friends I have (which authenticate with username and secret) also have an accountcode, and it shows up in the logs correctly. Is this normal behaviour? Does a friend without username/secret *not* log the accountcode when it places calls? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems compiling cdr_odbc.so
I'm not really being too lucky in the last days. After trying to compile cdr_mysql with no success, I am switching to cdr_odbc. I have installed unixODBC, iODBC and MyODBC correctly, I am even able to make queries with isql. But when trying to make in the cdr directory of the latest CVS, that's what I get: # cd /usr/src/asterisk/cdr # make cc -o cdr_odbc.so cdr_odbc.o -lodbc /usr/lib/gcc-lib/i386-redhat-linux/3.3.2/../../../crt1.o(.text+0x18): In function `_start': : undefined reference to `main' cdr_odbc.o(.text+0x197): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x1ae): In function `odbc_log': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x251): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x268): In function `odbc_log': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x5d9): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x5ed): In function `odbc_log': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x601): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x61b): In function `odbc_log': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x62d): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x647): In function `odbc_log': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x65c): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x670): In function `odbc_log': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x68d): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x6a1): In function `odbc_log': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x6ac): In function `odbc_log': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x6c0): In function `odbc_log': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x735): In function `odbc_unload_module': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x74f): In function `odbc_unload_module': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x7be): In function `odbc_unload_module': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x7d2): In function `odbc_unload_module': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x812): In function `odbc_unload_module': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x826): In function `odbc_unload_module': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x866): In function `odbc_unload_module': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x87a): In function `odbc_unload_module': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x8ba): In function `odbc_unload_module': : undefined reference to `option_verbose' cdr_odbc.o(.text+0x8ce): In function `odbc_unload_module': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0x904): In function `odbc_unload_module': : undefined reference to `ast_cdr_unregister' cdr_odbc.o(.text+0x95d): In function `odbc_load_module': : undefined reference to `ast_load' cdr_odbc.o(.text+0x993): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0x9ad): In function `odbc_load_module': : undefined reference to `ast_variable_browse' cdr_odbc.o(.text+0x9d7): In function `odbc_load_module': : undefined reference to `ast_variable_retrieve' cdr_odbc.o(.text+0xa4e): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0xa81): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0xaa9): In function `odbc_load_module': : undefined reference to `ast_variable_retrieve' cdr_odbc.o(.text+0xb20): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0xb53): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0xb7b): In function `odbc_load_module': : undefined reference to `ast_variable_retrieve' cdr_odbc.o(.text+0xbf2): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0xc25): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0xc4d): In function `odbc_load_module': : undefined reference to `ast_variable_retrieve' cdr_odbc.o(.text+0xcc6): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0xcef): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0xd2c): In function `odbc_load_module': : undefined reference to `ast_log' cdr_odbc.o(.text+0xd44): In function `odbc_load_module': : undefined reference to `ast_destroy' cdr_odbc.o(.text+0xd4d): In function `odbc_load_module': : undefined reference to `option_verbose' cdr_odbc.o(.text+0xd67): In function `odbc_load_module': : undefined reference to `ast_verbose' cdr_odbc.o(.text+0xd7f): In function `odbc_load_module': : undefined reference to `ast_verbose'
[Asterisk-Users] Unable to find libiodbc.so.2
I was finally able to compile asterisk with cdr_odbc.so. But now for some reason I get that error: *CLI load cdr_odbc.so Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource: libiodbc.so.2: cannot open shared object file: No such file or directory Unable to load module cdr_odbc.so But the file is there... # ls -lag /usr/local/lib/libiodbc.so* lrwxrwxrwx1 root 17 Jun 22 15:23 /usr/local/lib/libiodbc.so - libiodbc.so.2.1.9 lrwxrwxrwx1 root 17 Jun 22 15:23 /usr/local/lib/libiodbc.so.2 - libiodbc.so.2.1.9 -rwxr-xr-x1 root 1448547 Jun 22 15:23 /usr/local/lib/libiodbc.so.2.1.9 How do I tell cdr_odbc where to look for that file? These are all questions I couldn't find the answers to by looking anywhere... Am I the only one using cdr_odbc, or does it just plain work everywhere else? :) Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Re: cdr_addon_mysql compiling error
I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching the Wiki and the mailing list archive didn't bring up anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to switch back to 3.23? # make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: error: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: error: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: error: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: error: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 Two things: 1. Make sure you have the mysql-devel package installed, or equivalent. It is installed: # rpm -qa|grep MySQL MySQL-client-4.0.20-0 MySQL-devel-4.0.20-0 MySQL-shared-compat-4.0.18-0 MySQL-server-4.0.20-0 Do you know if MySQL 4 is supposed to work at all, or do I need to downgrade to 3.23? 2. Make sure you have done a make install in asterisk, before trying to do a make in asterisk-addons. Done that... Any ideas please? -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Re: cdr_addon_mysql compiling error
Thanks for the tip, but adding the CFLAGS directive doesn't work either, same error message. I'll try to have a look in -dev, but if anyone comes up with a solution, a reply would be appreciated. -Manuel -Messaggio originale- Da: Luckcuck Nick-LCKN001 [mailto:[EMAIL PROTECTED] Inviato: lunedì, 21. giugno 2004 13:52 A: [EMAIL PROTECTED] Oggetto: RE: [Asterisk-Users] Re: cdr_addon_mysql compiling error Hi, Another helpless person like me, I had the same problem a few days ago and a very helpful person suggested putting CFLAGS+=-I../asterisk/include in the Makefile, which worked fine for me. Maybe this is a problem with asterisk CVS ??? Don't really know but it might be worth looking at for some of you people in -dev, as a few people have asked this in the last few days. -- [ Nick Luckcuck | [EMAIL PROTECTED] ] [ Junior Software Developer | Motorola ] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Manuel Wenger Sent: 21 June 2004 12:43 To: [EMAIL PROTECTED] Subject: R: [Asterisk-Users] Re: cdr_addon_mysql compiling error I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching the Wiki and the mailing list archive didn't bring up anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to switch back to 3.23? # make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: error: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: error: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: error: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: error: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 Two things: 1. Make sure you have the mysql-devel package installed, or equivalent. It is installed: # rpm -qa|grep MySQL MySQL-client-4.0.20-0 MySQL-devel-4.0.20-0 MySQL-shared-compat-4.0.18-0 MySQL-server-4.0.20-0 Do you know if MySQL 4 is supposed to work at all, or do I need to downgrade to 3.23? 2. Make sure you have done a make install in asterisk, before trying to do a make in asterisk-addons. Done that... Any ideas please? -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Thousands of contexts?
By reading the Wiki's I found out that an Asterisk server with many (1) extensions and/or SIP users can become slow when reloading. But what happens when you also have many contexts in extensions.conf? More precisely, one context for each SIP user? I need this because I will have users with random usernames that they can choose, but I obviously cannot set that username as the outgoing caller ID when passing the call to our PSTN gateway. I need to change the CLI before dialling out. Now, every SIP user has his CLI, so I thought of creating a context for every user, where I would SetCallerID() before issuing the Dial() command. Obviously I would use some sort of script reading from a database to re-create the extensions.conf and sip.conf after making changes. Do you see any issues which could arise? Is Asterisk going to crash, or is it just going to be slow when reloading? Thank you for your help -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Thousands of contexts?
-Messaggio originale- Da: Kevin Walsh [mailto:[EMAIL PROTECTED] I don't quite understand your Caller*ID dilemma. In your sip.conf, you'd have a block for each user, say [abc123]. That's your random username, yes? The same block would also define the password and other directives. Why can't you simply include the callerid directive to set the Caller*ID name and number? You are right, of course... I forgot about the callerid directive and was looking at an impossible solution. Sorry about that... Guess I'm too tired today :-) -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_addon_mysql compiling error
I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching the Wiki and the mailing list archive didn't bring up anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to switch back to 3.23? # make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: error: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: error: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: error: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: error: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disable authentication on outgoing SIP calls
I am trying to make Asterisk communicate with a voice switch which doesn't need (and
like) authentication on outgoing SIP calls. I have configured it as follows in my
sip.conf:
[myswitch]
type=friend
host=192.168.1.100
port=5060
context=default
canreinvite=no
To dial out using this switch (it acts as a PSTN gateway) I use this in
extensions.conf:
exten = _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],90)
Incoming PSTN calls from myswitch work, Asterisk doesn't expect any authentication,
and doesn't get any, because the switch doesn't support it. Outgoing calls confuse the
switch, because Asterisk always wants to authenticate something, like this:
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6fe1dcea
From: My ATA sip:[EMAIL PROTECTED];tag=as0ff4afbb
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username=41911234567, realm=asterisk, algorithm=MD5,
uri=sip:[EMAIL PROTECTED], nonce=3135a7b3,
response=1cf43a75f985ca24a9f69ba785c2da23, opaque=
Date: Wed, 16 Jun 2004 17:24:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 295
The Proxy-Authorization part is what I need to remove from the INVITE request. Any
clues about how I could do that? I have already browsed Wikis and ML archives... any
help is appreciated
Thanks
-Manuel
___
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] hide caller id
Before starting to look at the problem in Asterisk, make sure that your phone company has enabled the selective CLIR feature. Otherwise the phone exchange will simply ignore your request to hide CLIP. Regards Manuel -Messaggio originale- Da: Pedro Vela [mailto:[EMAIL PROTECTED] Inviato: venerdì, 11. giugno 2004 08:56 A: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] hide caller id Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn´t work. What can I do, thaks Pedro ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] VoipTalk down?
Yes, same here. -Manuel -Messaggio originale- Da: Stephan Wik [mailto:[EMAIL PROTECTED] Inviato: venerdì, 11. giugno 2004 12:46 A: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] VoipTalk down? SIP registration not working, web site down. Anybody else see this? Stephan ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Hyperthreading?
That's the problem we had with Asterisk and HT on a 2.4 Kernel: whenever Asterisk was staying in the RTP stream, and HT was enabled (on a Dell Dual Xeon system), we had choppy audio. After disabling HT, everything was fine again. Nothing measurable, indeed, but you could definitely hear it. So there *must* be something. -Manuel -Messaggio originale- Da: Peter Corlett [mailto:[EMAIL PROTECTED] Inviato: martedì, 1. giugno 2004 16:09 A: [EMAIL PROTECTED] Oggetto: Re: [Asterisk-Users] Hyperthreading? Andrew Kohlsmith [EMAIL PROTECTED] wrote: [...] They can't? HT is detected in /proc/cpuinfo (flags) and I see two processors with 2.4.25 SMP kernels... What exactly isn't it using? Linux doesn't realise that scheduling a process onto one virtual CPU reduces the performance available on the other. There can be some quite bizarre scheduling decisions made as a result that can slow things down. On the other hand, for some tasks, it might not cause problems and thus you'll get a boost. As ever, the answer is to benchmark your configuration with and without HT to see. There's no simple answer. ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Hyperthreading?
We didn't have any card: we had choppy audio on SIP-to-SIP streams. There were no FXO/FXS cards in the system. I don't know what problem the other poster had. -Messaggio originale- Da: Chris Bond [mailto:[EMAIL PROTECTED] Inviato: martedì, 1. giugno 2004 16:22 A: [EMAIL PROTECTED] Oggetto: RE: [Asterisk-Users] Hyperthreading? What cards was it FXO - cos is it card related this HT problem? -Original Message- From: Manuel Wenger [mailto:[EMAIL PROTECTED] Sent: 01 June 2004 3:18 PM To: [EMAIL PROTECTED] Subject: R: [Asterisk-Users] Hyperthreading? That's the problem we had with Asterisk and HT on a 2.4 Kernel: whenever Asterisk was staying in the RTP stream, and HT was enabled (on a Dell Dual Xeon system), we had choppy audio. After disabling HT, everything was fine again. Nothing measurable, indeed, but you could definitely hear it. So there *must* be something. ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions/sip from database?
We are planning to deploy a pretty large asterisk server with many SIP extensions (might be up to 1 in the future), and I have a few questions: 1) is this possible, or are we running into some kind of limitation in the software that I wasn't aware of and that I didn't find by browsing through the archives and through Wiki? No, we don't need any G729-G711 transformations, it would only be acting as a SIP proxy (even if asterisk isn't a proxy). 2) is there a way to store extensions.conf and/or sip.conf in some kind of database, maybe MySQL? This would make life easier if someone wanted to change his SIP password. Or how would you otherwise solve this problem? 3) is there a quick way of reloading only a part of sip.conf/extensions.conf, for example if only a user password changed, or an extension's behaviour (eg. routing to voicemail instead of a SIP user)? Maybe I'm looking at the wrong software here and SER would be better for what I want to do... I know asterisk is supposed to be a PBX replacement, but the functions and flexibility it has really tells me stick with asterisk. Or am I way off with these assumptions? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DateTime bug?
I've just checked out the latest CVS from the 1.0-stable branch, but DateTime() seems somewhat buggy. It says something like: Tuesday May 18 11:46 AM 2004 instead of Tuesday May 18th 2004 at 11:46 AM (notice the wrong order of the words and the missing th/at) Did I miss something? Does DateTime() now take parameters that I wasn't aware of where you can tell * in what order it has to playback the date/time files? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?
Hi Tony, Try adding fromuser=x, maybe username= isn't enough... Just a guess, it already solved a few problems for me. -Manuel -Messaggio originale- Da: Tony Hoyle [mailto:[EMAIL PROTECTED] Inviato: martedì, 18. maggio 2004 13:03 A: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter? [...] [pipecall] type=peer secret=x username=x host=sipproxy.pipecall.com The first one works OK - I can dial out with no problems. The second one needs an extra field for the authuser - when I try to dial out I just get: May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to authenticate on INVITE to 'Tony Hoyle sip:[EMAIL PROTECTED];tag=as4afae981' I think this means it's using the wrong username somewhere... I can dial in just fine, so it's connected.. just only one way. Tony ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 on /dev/sda
I've just setup a new asterisk server, and I need to have G.729 working on this system. The problem is I don't have any IDE drives (and therefore no /dev/hda etc), but only /dev/sda. Is there really *no* way to license G.729 on a SCSI-only system? IMHO it's really stupid to replace an entire server because of a licensing issue. There *must* be a solution. Anyone, please? Or at least, is there anyone who knows who's the person (or the company) I should bother with this problem? Is it Digium or VoiceAge? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
