[asterisk-users] ${MACRO_CONTEXT} for Subroutines
Hello Everybody,
in past times I used macros but since a while they are deprecated.
So I replaced my macros with subroutines. In most cases this is really
no problem.
But in some rare cases I miss the macro channel variables (e.g.
${MACRO_CONTEXT}).
https://wiki.asterisk.org/wiki/display/AST/Dialplan+Macros+Channel+Variables
Is there something similar for subroutines?
Kind regards
Bastian
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[asterisk-users] Best way to connect analog modem
Hello Asterisk fans, I try to connect an analog modem to Asterisk. The modems are connected e.g. to alarm systems or a cash terminals (POS). As PSTN-Interface I'm using a Wildcard TE110P (E1). Is it possible to connect the modems to an ATA? Which ATA I should use for that scenarios? Cheers Bastian Virus checked by G DATA AntiVirusKit Version: AVK 17.1806 from 04.01.2007 Virus news: www.antiviruslab.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it possible to delete global variables
Tzafrir Cohen schrieb:
On Tue, May 16, 2006 at 01:09:41AM +0200, Bastian Schern wrote:
Hello,
is it possible to delete global variables during runtime?
Is setting the variable to an empty value good enough? How do you use
it?
I will use it to check if a Caller has already a open connection to a
specific destination.
In that moment I use e.g. Set(active_${CALLERIDNUM}=1|g) to set a global
variable in my dialplan. After the caller hung up I will use
Set(active_${CALLERIDNUM}=|g) to remove my global variable
"active_${CALLERIDNUM}".
Is the Memory space of variable really free? Or will I rise into a
memory problem if I do so?
Regards
Bastian
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[Asterisk-Users] Is it possible to delete global variables
Hello, is it possible to delete global variables during runtime? Regards Bastian Virus checked by G DATA AntiVirusKit Version: AVK 16.7382 from 15.05.2006 Virus news: www.antiviruslab.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delete global variable
Hello,
I've got a question to the memory handling of global variables in
Asterisk 1.2+.
I use e.g. Set(active_${CALLERIDNUM}=1|g) to set a global variable in my
dialplan. Later I will use Set(active_${CALLERIDNUM}=|g) to remove my
global variable "active_${CALLERIDNUM}".
Is the Memory space of variable really free?
Or will I rise into a memory problem?
Regards
Bastian
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[Asterisk-Users] chan_cap-cm-0.6 is not working for incomming calls
Hi, I tried to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it will not work: --- snip --- *CLI> capi debug CAPI Debugging Enabled -- CONNECT_IND (PLCI=0x101,DID=97,CID=0179903,CIP=0x1,CONTROLLER=0x1) == reventix: Incoming call '0179903' -> '97' -- reventix: info element CALLED PARTY NUMBER -- reventix: info element CHANNEL IDENTIFICATION 89 Urgent handler -- reventix: info element CALLED PARTY NUMBER -- reventix: info element CHANNEL IDENTIFICATION 89 Urgent handler == reventix: CAPI Hangingup == reventix: Interface cleanup PLCI=0x101 Urgent handler --- snap --- --- snip: capi.conf --- [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 ;ulaw=yes [reventix] ;ntmode=yes isdnmode=msn incomingmsn=* controller=1 group=1 ;prefix=0 softdtmf=on relaxdtmf=on accountcode= context=reventix-incoming holdtype=local ;immediate=yes ;echosquelch=1 ;echocancel=yes echocancelold=yes ;echotail=64 ;bridge=yes ;callgroup=1 ;deflect=1234567 devices=2 --- snap --- Asterisk is located behind a EURACOM PBX. With chan_capi-cm 0.5.4 it worked good. What's going wrong in my configuration? Regards Bastian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup with Dialog on snom display
Hello Everybody, I'm using the snom Phones together with Asterisk and I already able to see which Peer is used via "hint" priority. Then a LED on the snom phone is blinking. But I don't see who is calling the other phone. I know that the snom phones are already support this feature. But how I can enable this on Asterisk? Regards Bastian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapHFC E1 PRI (cwain)
Hello, I've got a Junghanns ZapHFC E1 PRI Card (cwain) and this driver writes very much messages into /var/log/messages like the following: --- snip --- Aug 2 17:58:02 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:02 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xef ] Aug 2 17:58:02 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:12 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:12 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xef ] Aug 2 17:58:12 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:22 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:22 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xef ] Aug 2 17:58:22 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:23 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x37 0x90 0xc3 ] 6 bytes Aug 2 17:58:23 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xef ] Aug 2 17:58:23 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 TX [ 0x0 0x1 0x36 0xee 0x8 0x2 0x0 0x68 0x5 0x4 0x3 0x80 0x90 0xa3 0x18 0x3 0xa9 0x83 0x81 0x6c 0xc 0x41 0x81 0x32 0x31 0x33 0x31 0x36 0x36 0x35 0x31 0x33 0x34 0x70 0xc 0xc1 0x36 0x34 0x34 0x31 0x31 0x37 0x31 0x39 0x31 0x39 0x30 0xa1 ] Aug 2 17:58:28 asterisk1 kernel: ztx 48 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 RX [ 0x0 0x1 0x1 0x38 0x11 0x2 ] 6 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xee 0x38 0x8 0x2 0x80 0x68 0x2 0x18 0x3 0xa9 0x83 0x81 0x1e 0x2 0x82 0x88 0x51 0x20 ] 20 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf0 ] Aug 2 17:58:28 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xf0 0x38 0x8 0x2 0x80 0x68 0x1 0xdd 0xf ] 11 bytes Aug 2 17:58:28 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf2 ] Aug 2 17:58:28 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xf2 0x38 0x8 0x2 0x80 0x68 0x45 0x8 0x2 0x84 0x91 0x1e 0x2 0x82 0x88 0xdc 0xc9 ] 19 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf4 ] Aug 2 17:58:31 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 TX [ 0x0 0x1 0x38 0xf4 0x8 0x2 0x0 0x68 0x4d 0x8 0x2 0x81 0x91 ] Aug 2 17:58:31 asterisk1 kernel: ztx 13 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xf4 0x3a 0x8 0x2 0x80 0x68 0x5a 0xab 0x84 ] 11 bytes Aug 2 17:58:31 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf6 ] Aug 2 17:58:31 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:41 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x3b 0xfc 0x9 ] 6 bytes Aug 2 17:58:41 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf7 ] Aug 2 17:58:41 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:51 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0x1 0x3b 0xfc 0x9 ] 6 bytes Aug 2 17:58:51 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf7 ] Aug 2 17:58:51 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 RX [ 0x2 0x1 0xf6 0x3a 0x8 0x2 0x26 0xb6 0x5 0x4 0x3 0x80 0x90 0xa3 0x18 0x3 0xa9 0x83 0x8c 0x6c 0x5 0x1 0x81 0x34 0x37 0x37 0x70 0x4 0x81 0x34 0x34 0x30 0x7d 0x2 0x91 0x81 0x9d 0x32 0x1 0x81 0xea 0xb2 ] 42 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 TX [ 0x2 0x1 0x1 0xf8 ] Aug 2 17:58:57 asterisk1 kernel: ztx 4 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 TX [ 0x0 0x1 0x3a 0xf8 0x8 0x2 0xa6 0xb6 0x2 0x18 0x3 0xa9 0x83 0x8c ] Aug 2 17:58:57 asterisk1 kernel: ztx 14 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 RX [ 0x0 0x1 0x1 0x3c 0x35 0x44 ] 6 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 TX [ 0x0 0x1 0x3c 0xf8 0x8 0x2 0xa6 0xb6 0x1 0x1e 0x2 0x81 0x88 ] Aug 2 17:58:57 asterisk1 kernel: ztx 13 bytes Aug 2 17:58:57 asterisk1 kernel: cwain: card 1 RX [ 0x0 0x1 0x1 0x3e 0x27 0x67 ] 6 bytes --- snap --- Is ist possible to disable this? Regards Bastian ___ 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] Suggestion for VoIP router with QoS
Hello, I'm searching for a router for our company. Does anybody has a suggestion for a router with a SIP Application Layer Gateway and good working QoS (Upstream AND Downstream). Regards Bastian ___ 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 and PostgreSQL
Hello everybody, now I'm using MySQL for SIP/IAX friends and CDR. Is it also possible to use PostreSQL instead of MySQL? Regards Bastian ___ 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 1.0.9 and PostreSQL DB
Hello everybody, now I'm using MySQL for SIP/IAX friends and CDR. Is it also possible to use PostreSQL instead of MySQL? Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting maximum runtime of echo test
Johnathan Corgan schrieb: Bastian Schern wrote: is it possible to limit the maximum runtime of the command "echo"? Use the AbsoluteTimeout application in your dialplan preceding the Echo application. http://voip-info.org/tiki-index.php?page=Asterisk%20AbsoluteTimeout -Johnathan Thanks allot Bastian ___ 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] Limiting maximum runtime of echo test
Hello everybody, is it possible to limit the maximum runtime of the command "echo"? Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: music on hold trouble
w fm3 schrieb: I too am having the same problem with =VS from last night. From my debugging, * never attempts to start MOH. Anyone else =ound this? Me too Music on hold - with SIP handsets at least - stopped working for me with asterisk 1.0.6 and cvs. If I downgraded to 1.0.5 works fine, upgrade and it stops working. all versions work fine if a dial an extension for music on hold. Cheers Walt Is there a solution for this Problem, I still have the same! Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
Tim De Lange schrieb: I'm using Asterisk, libpri and zaptel v1.0.6 on a SuSE Linux 9.1 with Kernel 2.6.9. I had not installed this Kernel, it is the default kernel on this dedicated server. As per README.SUSE in /usr/src/linux, try doing the following in /usr/src: make cloneconfig make modules_prepare Then change to zaptel directory and do: make clean make make install Hope this helps I made this already and it will not solve my problem. .-( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
Kristian Kielhofner schrieb: Bastian Schern wrote: If you look back in the archives, you will see that many, many, many people have gotten tripped up on the "make linux26" issue. Sorry to offend you. Remember that your original post never mentioned key details that would help. Speaking of the archives, perhaps you should do a search such as the following: http://www.google.com/search?q=invalid+module+format+site:lists.digium.com&hl=en&lr=&start=10&sa=N Also try the standard stuff - CVS head, stable, etc. Sorry, I described the problem with nearly no key details. I had already searched in Google for this Problem but I found no Answer there. I'm using Asterisk, libpri and zaptel v1.0.6 on a SuSE Linux 9.1 with Kernel 2.6.9. I had not installed this Kernel, it is the default kernel on this dedicated server. I had also tried to install a older version. Also no successs. On my local machine with SuSE 9.0 and Kernel 2.4 I have no Problems with v1.0.6. It seems to be a problem with the Kernel on the dedicated server. Does anybody has an idea how I can find out how to fix that problem. Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
Kristian Kielhofner schrieb: Bastian Schern wrote: Hello, I've got problems to install zaptel on a SuSE 9.1 System. The System has got a Linux 2.6.9 Kernel. If I try to load zaptel framework (modprobe zaptel) I get this message: FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format How I can fix this. At compile time, there were no Errors. Regards Bastian Did you do a "make linux26" in the zaptel directory? Yes, of course. Bastian ___ 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] FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
Hello, I've got problems to install zaptel on a SuSE 9.1 System. The System has got a Linux 2.6.9 Kernel. If I try to load zaptel framework (modprobe zaptel) I get this message: FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format How I can fix this. At compile time, there were no Errors. Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.6
Russell Bryant schrieb: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Greetings Everyone! Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the source as well as on the following web page: http://dev.asteriskdocs.org I had found this in the ChangeLogs: [...] -- chan_sip: [...] -- 'restrictcid' now properly works on MySQL peers. [...] Is there already DB-Support for sip.conf in this release? Or is it relating to ast_data? Regards Bastian ___ 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] Which version of ast_data for Asterisk v1.0.5?
Hi everybody, which version of ast_data I can use for Asterisk v1.0.5? Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] International area codes (incl. mobile)
PHP Mechanic schrieb: Hello everybody, does anybody knows from where I can get an list of international area codes incl. the mobile numbers? Have you tried google ? http://www.google.com.au/search?hl=en&q=international+dialing+codes Yes, I had tried that already. The search results containing no List with country codes and area codes including mobile. I only find the BT site, but they have no summarized table or something like that. There is only a query form. Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] International area codes (incl. mobile)
I'm interested in it. It would be very nice if you can send me your list. :-) But from where you got the Informations? Regards Bastian Sebastian Nocetti schrieb: I can send a list, mobile is not complete but it has a lot of numbers... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de PHP Mechanic Enviado el: Viernes, 07 de Enero de 2005 11:57 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] International area codes (incl. mobile) Hello everybody, does anybody knows from where I can get an list of international area codes incl. the mobile numbers? Have you tried google ? http://www.google.com.au/search?hl=en&q=international+dialing+codes ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 2005-01-06 ___ 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] International area codes (incl. mobile)
Hello everybody, does anybody knows from where I can get an list of international area codes incl. the mobile numbers? Regards Bastian ___ 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] Problem with Accounting and wrong Caller ID
Hi everybody, how I can ensure that the Authorization-ID instead of the CID is logged to the Accounting Database? E.g. if somebody enters a fake CID number an name this fake IDs are logged to the Accounting database. Now it is not possible to identity the real caller. Regards Bastian ___ 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
Re: [Asterisk-Users] no plain text passwords in iax.conf
Adam Hart schrieb: Bastian Schern wrote: Adam Hart schrieb: Bastian Schern wrote: Hello Asterisk friends, is it possible to avoid plain text passwords in the iax.conf or the iaxfriends MySQL database table? Asterisk needs the plain text password to authenicate. You could wrap a base64 decode when reading the passwords, but this is obsecurity, yet simple to implement & should prevent the casual browser. I guess a more secure method would public key crypto and give asterisk the key at runtime (obviously not 100% secure either) I found out that MySQL offers some methods to store strong passwords: http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers But how I use this with Asterisk? That's using private key crypto, when you store the password you do aes_encode(password,"somekey") then when asterisk reads it, do a aes_decode(password,"somekey") - modify chan_iax2 when you do the select - change the SQL statement: the column 'secret' to 'aes_decode(secret,"somekey") as real_secret' then below change secret to real_secret. What is about the field md5secret similar to sip.conf? Is that not a solution for iax.conf? Bastian ___ 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
Re: [Asterisk-Users] no plain text passwords in iax.conf
Adam Hart schrieb: Bastian Schern wrote: Hello Asterisk friends, is it possible to avoid plain text passwords in the iax.conf or the iaxfriends MySQL database table? Asterisk needs the plain text password to authenicate. You could wrap a base64 decode when reading the passwords, but this is obsecurity, yet simple to implement & should prevent the casual browser. I guess a more secure method would public key crypto and give asterisk the key at runtime (obviously not 100% secure either) I found out that MySQL offers some methods to store strong passwords: http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers But how I use this with Asterisk? Bastian ___ 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] no plain text passwords in iax.conf
Hello Asterisk friends, is it possible to avoid plain text passwords in the iax.conf or the iaxfriends MySQL database table? Regards Bastian ___ 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] advise for cheap ISDN card which works with chan_capi and supports p2p mode
Hi everybody, does anybody has an advise for a cheap ISDN card which works with chan_capi and supports p2p mode. Regards Bastian ___ 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] P2P (DDI) mode with chan_capi 0.3.5
Hi everybody,
I have problems with the chan_capi in the P2P (DDI) mode and it is not
possible to use zaphfc, because I need call defelction.
This is my capi.conf:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de
;mode=immediate
isdnmode=ptp
; 123456 -0
[interfaces]
msn=1234560
incomingmsn=1234560
controller=1
devices=2
;mode=immediate
isdnmode=ptp
softdtmf=1
accountcode=
context=AH-P
echocancel=yes
If I dial a number over the CAPI I get this:
--- snip ---
parrot*CLI> capi debug
CAPI Debugging Enabled
-- Executing Dial("SIP/40-fecc", "CAPI/1234560:B08003301000)") in
new stack
-- data = 1234560:B08003301000)
-- capi request omsn = 1234560
== found capi with omsn = 1234560
== CAPI Call CAPI[contr1/1234560]/1 with B3-- CONNECT_CONF ID=002
#0x0005 LEN=0014
Controller/PLCI/NCCI= 0x101
Info= 0x0
== received CONNECT_CONF PLCI = 0x101 INFO = 0
-- Called 1234560:B08003301000)
-- DISCONNECT_IND ID=002 #0x000d LEN=0014
Controller/PLCI/NCCI= 0x101
Reason = 0x3302
== DISCONNECT_IND PLCI=0x101 REASON=0x3302
== No one is available to answer at this time
Nov 26 15:16:27 NOTICE[344084]: rtp.c:429 ast_rtp_read: RTP: Received
packet with bad UDP checksum
-- Timeout on SIP/40-fecc
== CDR updated on SIP/40-fecc
-- Executing Goto("SIP/40-fecc", "default|s|1") in new stack
-- Goto (default,s,1)
-- Executing Hangup("SIP/40-fecc", "") in new stack
== Spawn extension (default, s, 1) exited non-zero on 'SIP/40-fecc'
--- snap ---
Does Anybody knows, what is going wrong?
Regards
Bastian
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[Asterisk-Users] No debugging informations on the CLI after patching with ast_data 1.0.2
Hi to everybody,
I have the problem that nearly no information are displayed on the
Asterisk CLI (asterisk -r). In former times (before patching Asterisk
1.0.2 with ast_data 1.0.2) it looks e.g. like this:
--- snip ---
-- Registered '96' (AUTHENTICATED) at 212.202.169.118:4569
-- Accepting AUTHENTICATED call from 212.202.169.118, requested
format = 1024, actual format = 1024
-- Executing Macro("IAX2/[EMAIL PROTECTED]/2", "echo") in new stack
-- Executing Playback("IAX2/[EMAIL PROTECTED]/2", "demo-echotest") in new stack
-- Playing 'demo-echotest' (language 'de')
== Spawn extension (macro-echo, s, 1) exited non-zero on
'IAX2/[EMAIL PROTECTED]/2' in macro 'echo'
== Spawn extension (imatris, 600, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/2'
-- Executing Goto("IAX2/[EMAIL PROTECTED]/2", "default|s|1") in new stack
-- Goto (default,s,1)
-- Executing Hangup("IAX2/[EMAIL PROTECTED]/2", "") in new stack
== Spawn extension (default, s, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/2'
-- Hungup 'IAX2/[EMAIL PROTECTED]/2'
--- snap ---
Now I see nothing of that.
Does anybody has the same problem or know how to get back the former
behaviour?
Regards
Bastian
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[Asterisk-Users] Asterisk on a Linksys WRT54G(S)
Hello to everybody, does anybody knows how to install Asterisk on a Linksys WRT54G(S)? I had read in the Wiki that it is possible. If somebody has a tip, this would help me very much. Regards Bastian ___ 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
Re: [Asterisk-Users] Call Deflection (CD) with ZapHFC
Does nobody know whether this is possible or not? Bastian Schern schrieb: Hi to everybody, is it possible to use ISDN Call Deflection with a ZapHFC card? Regards Bastian ___ 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 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
Re: [Asterisk-Users] IAX2 peers via MySQL DB with Asterisk 1.0.2
This works fine for me, thanks! But a after the installation the *CLI will not show any longer what's going on. How comes this? Regards Bastian Michael Shuler schrieb: Use this http://svn.asteriskdocs.org/res_data/ It will get you sipfriends, IAX, etc. all live from MySQL/ODBC/etc. Michael Shuler, C.E.O. BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP) 682 High Point Lane East Peoria, IL 61611 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: [EMAIL PROTECTED] Customer Service: (877) 976-0711 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bastian Schern Sent: Thursday, November 18, 2004 8:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 peers via MySQL DB with Asterisk 1.0.2 Does nobody else has got this Problem? Or does nobody know how it should be fixed? ;-) Bastian Schern schrieb: Hi all, I try to get the IAX2 peers from a MySQL DB. But this will not work. I'm working with Asterisk 1.0.2. First I had enabled MYSQL_FRIENDS in the channels Makefile, then I had created the user "asterisk", the database "asterisk" and the table "iaxfriends" inside my MySQL server (modelled on http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers). I add this to my iax.conf: --- snip --- dbname=asterisk ; Name of database in your MySQL server dbhost=localhost; Hostname of server dbuser=asterisk ; Username in MySQL dbpass=123456 ; Password for user in MySQL --- snap --- A added the User as following: mysql> INSERT INTO `iaxfriends` ( `name` , `secret` , `context` , `ipaddr` , `port` , `regseconds` ) VALUES ( 'bastian', '123456', 'default', '', '0', '0' ); After that I start my Asterisk without problems, but the peers defined in the "iaxfriends" table are not working': SP2106*CLI> iax2 show peers Name/UsernameHost Mask Port Status If a IAX-Client tries to connect it will produce this message: --- snip --- -- Unregistered 'bastian' (AUTHENTICATED) --- snap --- If I try to dial from the client I will get this on the Asterisk console: --- snip --- Nov 16 20:30:00 NOTICE[1092774832]: chan_iax2.c:5402 socket_read: Rejected connect attempt from 212.202.169.118 --- snap --- But inside the DB-Table it looks like the client is registered: mysql> SELECT * FROM iaxfriends; +-++-+-+--++ | name| secret | context | ipaddr | port | regseconds | +-++-+-+--++ | bastian | r2dzwo | default | 212.202.169.118 | 4569 | 1100655488 | +-++-+-+--++ 1 row in set (0.00 sec) What is wrong? Regards Bastian ___ 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 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 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 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] Call Deflection (CD) with ZapHFC
Hi to everybody, is it possible to use ISDN Call Deflection with a ZapHFC card? Regards Bastian ___ 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
Re: [Asterisk-Users] IAX2 peers via MySQL DB with Asterisk 1.0.2
Does nobody else has got this Problem? Or does nobody know how it should be fixed? ;-) Bastian Schern schrieb: Hi all, I try to get the IAX2 peers from a MySQL DB. But this will not work. I'm working with Asterisk 1.0.2. First I had enabled MYSQL_FRIENDS in the channels Makefile, then I had created the user "asterisk", the database "asterisk" and the table "iaxfriends" inside my MySQL server (modelled on http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers). I add this to my iax.conf: --- snip --- dbname=asterisk ; Name of database in your MySQL server dbhost=localhost; Hostname of server dbuser=asterisk ; Username in MySQL dbpass=123456 ; Password for user in MySQL --- snap --- A added the User as following: mysql> INSERT INTO `iaxfriends` ( `name` , `secret` , `context` , `ipaddr` , `port` , `regseconds` ) VALUES ( 'bastian', '123456', 'default', '', '0', '0' ); After that I start my Asterisk without problems, but the peers defined in the "iaxfriends" table are not working': SP2106*CLI> iax2 show peers Name/UsernameHost Mask Port Status If a IAX-Client tries to connect it will produce this message: --- snip --- -- Unregistered 'bastian' (AUTHENTICATED) --- snap --- If I try to dial from the client I will get this on the Asterisk console: --- snip --- Nov 16 20:30:00 NOTICE[1092774832]: chan_iax2.c:5402 socket_read: Rejected connect attempt from 212.202.169.118 --- snap --- But inside the DB-Table it looks like the client is registered: mysql> SELECT * FROM iaxfriends; +-++-+-+--++ | name| secret | context | ipaddr | port | regseconds | +-++-+-+--++ | bastian | r2dzwo | default | 212.202.169.118 | 4569 | 1100655488 | +-++-+-+--++ 1 row in set (0.00 sec) What is wrong? Regards Bastian ___ 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 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] Removed default indication country 'us'
Hi all, what is the meaning of this message: Nov 17 19:18:27 NOTICE[514032]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' Regards Bastian ___ 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] IAX2 peers via MySQL DB with Asterisk 1.0.2
Hi all, I try to get the IAX2 peers from a MySQL DB. But this will not work. I'm working with Asterisk 1.0.2. First I had enabled MYSQL_FRIENDS in the channels Makefile, then I had created the user "asterisk", the database "asterisk" and the table "iaxfriends" inside my MySQL server (modelled on http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers). I add this to my iax.conf: --- snip --- dbname=asterisk ; Name of database in your MySQL server dbhost=localhost; Hostname of server dbuser=asterisk ; Username in MySQL dbpass=123456 ; Password for user in MySQL --- snap --- A added the User as following: mysql> INSERT INTO `iaxfriends` ( `name` , `secret` , `context` , `ipaddr` , `port` , `regseconds` ) VALUES ( 'bastian', '123456', 'default', '', '0', '0' ); After that I start my Asterisk without problems, but the peers defined in the "iaxfriends" table are not working': SP2106*CLI> iax2 show peers Name/UsernameHost Mask Port Status If a IAX-Client tries to connect it will produce this message: --- snip --- -- Unregistered 'bastian' (AUTHENTICATED) --- snap --- If I try to dial from the client I will get this on the Asterisk console: --- snip --- Nov 16 20:30:00 NOTICE[1092774832]: chan_iax2.c:5402 socket_read: Rejected connect attempt from 212.202.169.118 --- snap --- But inside the DB-Table it looks like the client is registered: mysql> SELECT * FROM iaxfriends; +-++-+-+--++ | name| secret | context | ipaddr | port | regseconds | +-++-+-+--++ | bastian | r2dzwo | default | 212.202.169.118 | 4569 | 1100655488 | +-++-+-+--++ 1 row in set (0.00 sec) What is wrong? Regards Bastian ___ 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
Re: [Asterisk-Users] UDP Fragmentation Problem
Rich Adamson schrieb: As far as I am aware there is no such thing as a fragmented UDP packet; each packet is sent out on its own, there is no coherency between UDP packets like there is with TCP packets. I could be very wrong here, it's been a late night with the kids. :-) Packet fragmentation is at the IP layer, so UDP will have fragmented packets too. But... the OS should handle that and Asterisk shouldn't find out - it's a all or none policy, so it should receive the whole packet at once or nothing. How I can setup Linux to handle UDP fragments? Not sure why the concern with fragmentation, it should not be an issue with any modern linux distribution and there is nothing to setup. The only issue that I've heard about in recent months/years relative to fragmentation is the SonicWall firewall just can't seem to get it right. In their case, any udp packet greater then about 1500 bytes does not get reassembled propery, and its still an issue in the latest firmware. If you really think you've got a fragmentation problem, I'd like to see a packet trace (eg, ethereal) of those packets. Here it is ;-) Okay, looked at the pcap and see the fragmentation, but that does not indicate your asterisk IP stack is not handling it properly. Might compare a 'sip debug' with those packets to see if data is reassembled. Since both pieces of the original fragmented packet did in fact arrive at your destination, the only issue left is whether your IP stack reassembled them properly. I'd suspect another problem is lurking unrelated to fragmentation. I think you're right it seems to be the client side in Colombia. Tomorrow I will perform a trace there. You've got a idea problem it could be? Regards Bastian Rich ___ 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 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
Re: [Asterisk-Users] UDP Fragmentation Problem
Rich Adamson schrieb: I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? As far as I am aware there is no such thing as a fragmented UDP packet; each packet is sent out on its own, there is no coherency between UDP packets like there is with TCP packets. I could be very wrong here, it's been a late night with the kids. :-) Packet fragmentation is at the IP layer, so UDP will have fragmented packets too. But... the OS should handle that and Asterisk shouldn't find out - it's a all or none policy, so it should receive the whole packet at once or nothing. How I can setup Linux to handle UDP fragments? Not sure why the concern with fragmentation, it should not be an issue with any modern linux distribution and there is nothing to setup. The only issue that I've heard about in recent months/years relative to fragmentation is the SonicWall firewall just can't seem to get it right. In their case, any udp packet greater then about 1500 bytes does not get reassembled propery, and its still an issue in the latest firmware. If you really think you've got a fragmentation problem, I'd like to see a packet trace (eg, ethereal) of those packets. Here it is ;-) Regards Bastian javier-sip-1.pcap Description: Binary data ___ 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
Re: [Asterisk-Users] UDP Fragmentation Problem
Adam Hart schrieb: Andrew Kohlsmith wrote: On October 31, 2004 05:36 pm, Bastian Schern wrote: I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? As far as I am aware there is no such thing as a fragmented UDP packet; each packet is sent out on its own, there is no coherency between UDP packets like there is with TCP packets. I could be very wrong here, it's been a late night with the kids. :-) Packet fragmentation is at the IP layer, so UDP will have fragmented packets too. But... the OS should handle that and Asterisk shouldn't find out - it's a all or none policy, so it should receive the whole packet at once or nothing. How I can setup Linux to handle UDP fragments? ___ 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
Re: [Asterisk-Users] UDP Fragmentation Problem
Rich Adamson schrieb: Bastian Schern wrote: Hi everybody, I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? Fragmentation should not matter for the end-point (the source or destination of the UDP datagram), since the IP stack itself should take care of the reassembly.. Butit is quite weird they have such a small MTU. Many websites that have problems with Path MTU discovery would be broken by that (dumb websites, but still, way too many...). Fragmentation shouldn't make any difference as the sip/rtp/g711 packets are roughly 250 bytes anyway. In the case of SIP is imho not correct. E.g. a SIP REGISTER packet is round about 656 Bytes long. RTP I don't know. And what is with RTP-GSM/G729 packets? Regards Bastian ___ 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] UDP Fragmentation Problem
Hi everybody, I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? Regards Bastian ___ 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] SER + Asterisk
Hi, since a while I try get Asterisk and SER work together. But until now I have no success. I want to use Asterisk as Gateway to the old telephone world. Is there somebody who can give me a small example of the ser.cfg and the Asterisk config files. This will be very nice. Thanks Bastian ___ 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
Re: [Asterisk-Users] Snom 200 updates
WipeOut schrieb: I always just let the phone poll the Snom update server for updates but while the server is back at version 2.03o the latest stable downloadable version on the website is 2.04n.. Is Snom not distributing updates for the 200 from their server anymore?? Have a look here: http://www.snom.com/download/share/ Regards Bastian ___ 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] Incoming MSN via ZapHFC -> to SIP
Hi there,
I've got a small problem with the zaphfc channel. No MSN of an any
incoming call which comes trough the ISDN card (Acer ISDN, with HFC
chipset and zaphfc driver) which will be forwarded to the SIP-Phone will
be displayed. Always it will be shown "asterisk" an the Display.
--- snip (zapata.conf) ---
[channels]
language=de
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan=local
prilocaldialplan=local
pritrustusercid = yes
echocancel=yes
immediate=no
group = 1
context=default
channel => 1-2
--- snap ---
--- snip (extensions.conf) ---
[general]
static=yes
writeprotect=yes
[globals]
BASTIAN=SIP/16
[macro-callwithmsn]
exten => s,1,SetCallerID(${ARG2})
exten => s,2,SetCIDName(${ARG3})
exten => s,3,Dial(Zap/g1/${ARG1},60,Ttr)
exten => s,104,Playtones(busy);
exten => s,105,Busy
[default]
exten => 96,1,SetCIDNum(${CALLERIDNUM})
exten => 96,2,Dial(SIP/16)
exten => _0.,1,Macro(callwithmsn,${EXTEN:1},61,Bastian)
exten => _XX,1,Dial(SIP/${EXTEN})
--- snap ---
It would be very nice if somebody can help me.
Regards
Bastian
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[Asterisk-Users] ZAPHFC for Euro-ISDN
Hello *, I had successfully downloaded and installed Asterisk via bri-stuff.0.1.0-RC3. I want to drive the ZAPHFC-Card in German Euro-ISDN TE-Mode but actually I have no idea how to setup the MSNs for that device. Can somebody show me a example of extension.conf, zaptel.conf and zapata.conf for e.g. the MSNs 12345671 up to 12345679. This would be very nice. Regards Bastian ___ 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
Re: [Asterisk-Users] 2.4.x-SMP vs. 2.6.x-SMP
Jean-Yves Avenard schrieb: Hello 2.6 scheduler performs in O(1), it will perform much better in multi-processor environment than the 2.4 series That's one thing, but what is with the compatibility? CAPI? ZapHFC? And so on. Regards Bastian ___ 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] 2.4.x-SMP vs. 2.6.x-SMP
Hi *, I want start with a setup of Asterisk with a clean PC. This PC is a SMP-Machine with two 466MHz CPUs, a Acer ISDN card and a AVM Fritz! PCI card. Which Kernel is better for my constellation (Asterisk with SMP, CAPI and ZAPHFC)? Kernel 2.6.x or Kernel 2.4.x? Regards Bastian ___ 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] Asterisk in a DMZ
Hello *, I try to establish a Asterisk-Server for internal and external usage. Perfect use case for a DMZ, or not? My configuration: I N T E R N E T | | | E | | X | | T | | E | 213.xxx.xx.68 | R +-#+| N | Firewall || +-#+ - - - - - - - - - - - - - - - - - - - -+- | 192.168.40.68 | | | +#+| | Switch || +--#---#---#---#--+| | | | | +-+ | D | | | M +--+ | | Z | (213.xxx.xx.66) | (213.xxx.xx.70) | | 192.168.40.66| 192.168.40.70 | +-#+ +-#+| | Firewall | | Asterisk || +--+ +--+| | Server | | +-#+ - - - - - - - - - - - - - - - - - - - - -+- | 192.168.0.1| || +--+ | | | +#+| | Switch || I +--#--#--#--#--#--+| N | | | | T | | | | E | | | | R | | | | N | | +-+ | | +--+ | | | | | | | 192.168.0.101 | 192.168.0.102 | 192.168.0.103 | +--#---+ +--#---+ +--#---+ | | Tel1 | | Tel2 | | Tel3 | | +--+ +--+ +--+ | But now the IP-Phones could not communicate with Asterisk because the Server (a Linux host) will NAT the internal IP-Addresses. Is there a good way to solve this Problem? Regards Bastian ___ 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] German sounds
Hi *, are there already some free German sounds for Asterisk? Regards Bastian ___ 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
Re: [Asterisk-Users] Problems with variables
Seth Remington schrieb:
On Tue, 2004-07-27 at 09:40, Bastian Schern wrote:
Hi *,
I have problems with the variables in the extensions.conf file.
--- snip ---
[global]
JOHN=SIP/17
[incoming]
exten => s,1,Answer()
exten => s,2,Playback(demo-enterkeywords)
exten => s,3,Background(demo-congrats)
exten => 1,1,Dial(SIP/17)
exten => 2,1,Dial(${JOHN})
--- snap ---
If I dial 1 it will work fine, but if I dial 2 I will get this Message
in the Asterisk CLI:
Jul 27 17:39:34 WARNING[425999]: app_dial.c:485 dial_exec: Dial argument
takes format (technology1/number1&technology2/number2...|optional timeout)
[global] should be [globals]
-Seth
Ups,
I got my information from here:
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x680.html
Is it an error in the doc? ;-)
thanks for help
Bastian
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[Asterisk-Users] Problems with variables
Hi *,
I have problems with the variables in the extensions.conf file.
--- snip ---
[global]
JOHN=SIP/17
[incoming]
exten => s,1,Answer()
exten => s,2,Playback(demo-enterkeywords)
exten => s,3,Background(demo-congrats)
exten => 1,1,Dial(SIP/17)
exten => 2,1,Dial(${JOHN})
--- snap ---
If I dial 1 it will work fine, but if I dial 2 I will get this Message
in the Asterisk CLI:
Jul 27 17:39:34 WARNING[425999]: app_dial.c:485 dial_exec: Dial argument
takes format (technology1/number1&technology2/number2...|optional timeout)
Does anybody knows where the problem is?
Cheers
Bastian
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