[Asterisk-Users] asterisk realtime msql
Hi there
asterisk goes to 90% cpu usage when trying to authenticate a sip friend using
realtime mysql, no other message does appear at cli and asterisk hungs;
here some info:
*CLI realtime load sipfriends name 104
Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:109 realtime_mysql: MySQL
RealTime: Retrieve SQL: SELECT * FROM sipfriends WHERE name = '104'
Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:637 mysql_reconnect: MySQL
RealTime: Everything is fine.
Column Name Column Value
uniqueid 1
name 104
accountcode 104
callgroup 1
context default
host dynamic
secret abc
type friend
username 104
regseconds 0
*CLI realtime mysql status
Jan 13 11:57:06 DEBUG[8928]: res_config_mysql.c:637 mysql_reconnect: MySQL
RealTime: Everything is fine.
Connected to [EMAIL PROTECTED], port 3306 with username root for 4 seconds.
in extconfig.conf i have:
[settings]
;family name = driver,database name[,table_name]
sipfriends = mysql,asterisk,sipfriends
in res_mysql.conf
[general]
dbhost = 192.168.1.10
dbname = asterisk
dbuser = root
dbpass =
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
thnx in advance
any help will be very apreciated :)
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Maurizio Marini
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Re: [Asterisk-Users] CAPI and Asterisk (with AVM ISDN Card)
On Wednesday 20 October 2004 00:30, Mateo Meier wrote: Does anybody knows what version of capi is needed ? try the most recent here: ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots it did work fine for me (FC2 and debian sid) I tried to install a capi rpm.. but after the capi rpm installation, there seems to be no /etc/capi.conf cat capi.conf # card fileproto io irq mem cardnr options b1isa b1.t4 DSS10x150 7 - - P2P b1pci b1.t4 DSS1- - - - c4 /usr/sbin/c4.binDSS1- - - - c4 - DSS1- - - - c4 - DSS1- - - - P2MP c4 - DSS1- - - - P2MP c2 c2.bin DSS1- - - - c2 - DSS1- - - - t1isa t1.t4 DSS10x340 9 - 0 t1pci t1.t4 DSS1- - - - fcpci - - - - - - fcclassic - - 0x150 10 - - What kind of capi version do I need ? capi4k-utils ? or just any capi rpm ? download a tarball and install it... Maurizio ___ 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] spandsp / fax partially received
at last (using libtiff 3.5.7 as Mike Machado suggested) i was able to get spandsp working on my debian sid 2.6.8.1, with * up to 18-08-2004 and bristuff 4a i've tested only fax receiving and the file on /var/spool/asterisk-fax/ contains only the first 3-4 cm. of page sent sending fax notified an 'ok' fax status Maurizio ___ 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] more on spandsp and partially received fax
). Fax3Decode2D: (FakeInput): Bad code word at scanline 273 (x 1289). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 273 (got 1289, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 274 (got 1757, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 277 (got 1745, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 278 (got 3531, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 279 (x 1430). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 279 (got 1430, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 280 (got 1735, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 281 (got 2246, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 282 (x 964). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 282 (got 964, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 283 (got 1734, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 284 (got 1729, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 285 (got 1729, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 287 (got 1735, expected 1728). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 289 (got 0, expected 1728). Page 1 of /var/spool/asterisk-fax/1095774866.2.tif: 290 rows received 0 total bad rows 0 max consecutive bad rows Changed from phase 5 to 3 Slow carrier up EOP: 2f EOP with final frame tag In state 5 Changed from phase 3 to 4 MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Urgent handler Slow carrier up DCN: fb DCN with final frame tag In state 8 Disconnecting Changed from phase 3 to 7 Sep 21 15:55:08 DEBUG[1120357296]: app_rxfax.c:66 phase_e_handler: == Sep 21 15:55:08 DEBUG[1120357296]: app_rxfax.c:67 phase_e_handler: Fax successfully received. Sep 21 15:55:08 DEBUG[1120357296]: app_rxfax.c:68 phase_e_handler: Remote station id: Sep 21 15:55:08 DEBUG[1120357296]: app_rxfax.c:69 phase_e_handler: Local station id: Sep 21 15:55:08 DEBUG[1120357296]: app_rxfax.c:70 phase_e_handler: Pages transferred: 1 Sep 21 15:55:08 DEBUG[1120357296]: app_rxfax.c:71 phase_e_handler: Resolution:1 Sep 21 15:55:08 DEBUG[1120357296]: app_rxfax.c:72 phase_e_handler: Transfer Rate: 9600 Sep 21 15:55:08 DEBUG[1120357296]: app_rxfax.c:73 phase_e_handler: == Changed from phase 7 to 8 -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 ___ 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] spandsp
On Sunday 19 September 2004 00:46, Edward Eastman wrote:
I think the port.h in this distribution may have been created from
tiffv3.5.7 while you have tiffv3.6.0 - (or maybe something else), anyway I
had this problem, and installing tiffv3.5.7 and copying the port.h from
that distribution to /usr/local/include fixed it
i solved the issue commenting the declaration in tif_dir.h
--- tif_dir.h~ 2004-09-17 13:31:40.0 +0200
+++ tif_dir.h 2004-09-17 13:31:40.0 +0200
@@ -228,16 +228,16 @@
((v) (tif)-tif_typemask[type]) (tif)-tif_typeshift[type] : \
(v) (tif)-tif_typemask[type]))
-typedefstruct {
- ttag_t field_tag; /* field's tag */
- short field_readcount;/* read count/TIFF_VARIABLE/TIFF_SPP */
- short field_writecount; /* write count/TIFF_VARIABLE */
- TIFFDataType field_type;/* type of associated data */
- u_short field_bit; /* bit in fieldsset bit vector */
- u_char field_oktochange; /* if true, can change while writing */
- u_char field_passcount;/* if true, pass dir count on set */
- char*field_name;/* ASCII name */
-} TIFFFieldInfo;
+//typedef struct {
+// ttag_t field_tag; /* field's tag */
+// short field_readcount;/* read count/TIFF_VARIABLE/TIFF_SPP */
+// short field_writecount; /* write count/TIFF_VARIABLE */
+// TIFFDataType field_type;/* type of associated data */
+// u_short field_bit; /* bit in fieldsset bit vector */
+// u_char field_oktochange; /* if true, can change while writing */
+// u_char field_passcount;/* if true, pass dir count on set */
+// char*field_name;/* ASCII name */
+//} TIFFFieldInfo;
#defineTIFF_ANYTIFF_NOTYPE /* for field descriptor searching */
#defineTIFF_VARIABLE -1 /* marker for variable length tags */
about the other issue i got:
[app_rxfax.so]Sep 17 18:14:33 WARNING[1076992544]: loader.c:242 ast_load_resource:
/usr/lib/asterisk/modules/app_rxfax.so: symbol errno, version GLIBC_2.0 not defined in
file libc.so.6 with link time reference
i solved it grepping errno into apps dir and noting that all the other sources don't
have any decl. as app_rxfax and app_txfax:
extern int errno;
all of them have:
#include errno.h
so i add this include and i removed external reference:
--- app_txfax.c~2004-09-20 10:32:00.0 +0200
+++ app_txfax.c 2004-09-20 10:32:00.0 +0200
@@ -22,6 +22,7 @@
#include asterisk/translate.h
#include string.h
#include stdlib.h
+#include errno.h
#include stdint.h
#include pthread.h
#include tiffio.h
@@ -90,7 +91,7 @@
uint8_t __buf[sizeof(uint16_t)*BLOCK_SIZE + 2*AST_FRIENDLY_OFFSET];
uint8_t *buf = __buf + AST_FRIENDLY_OFFSET;
int len;
-extern int errno;
+// extern int errno;
if (chan == NULL)
{
maurizio
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Re: [Asterisk-Users] spandsp
On Monday 20 September 2004 14:49, Rich Adamson wrote: Are you applying spandsp against a recent * cvs? no, i use bri_stuff last ver, it's up to 18-08-2004 Maurizio ___ 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] spandsp core dumps asterisk receiving fax
debian sid with 2.6.8.1 kernel at last i was able to get spandsp compiled (see my last post) now i try to receive a fax but ...core dump!! :( here the output at console (nothing under /var/log/asterisk): Changed from phase 0 to 1 Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1712.90 (88) Fast carrier down Fast carrier up Coarse carrier frequency 1700.09 (66) Training error 7.764798 Training succeeded (constellation mismatch 6.026207) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Coarse carrier frequency 1699.83 (66) Training error 6.394423 Training succeeded (constellation mismatch 10.554763) Fast carrier trained Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) Warning, flexibel rate not heavily tested! some guru can help? -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 ___ 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] spandsp / compilation errors
On Monday 20 September 2004 18:54, Steve Underwood wrote: I would think its because it cannot find libspandsp, wouldn't you? Is it installed? Is it installed in your library path? Like most things built with the ./configure/make make install sequence, the default is for the library to go in /usr/local/lib. Is that where it is? Is that in your library path? try ./configure /usr to get rid of it all the stuff will be created under /usr instead of /usr/local -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 ___ 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] GrandStream BT101 Attended Transfers
On Friday 20 August 2004 09:54, Massimo De Nadal wrote: I've asked Grandstream tech support about attended transfer. They told me that in about a month there will be available a firmware upgrade that supports attended transfer natively. maxx any news? -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 ___ 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] spandsp
On Thursday 19 August 2004 23:29, administrator tootai wrote: I made one. Can be found at http://ftp2.tootai.net/spandsp-0.0.1k-whole.tar.gz The 3 headers files are included, made a short readme file for installation and modify the Makefile.patch (remove the dtmftotext). Comments welcome. debian sid with littiff3-dev libtiff4-dev installed; compiling spandsp i get this error: gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t4.c -MT t4.lo -MD -MP -MF .deps/t4.TPlo -fPIC -DPIC -o .libs/t4.lo In file included from /usr/include/tiffiop.h:45, from t4.c:38: /usr/include/tif_dir.h:240: error: conflicting types for `TIFFFieldInfo' /usr/include/tiffio.h:448: error: previous declaration of `TIFFFieldInfo' make[2]: *** [t4.lo] Error 1 `TIFFFieldInfo' is defined in tif_dir.h and in my tiffio.h: /usr/include# grep TIFFFieldInfo * tif_dir.h:} TIFFFieldInfo; tif_dir.h:externvoid _TIFFMergeFieldInfo(TIFF*, const TIFFFieldInfo[], int); tif_dir.h:externconst TIFFFieldInfo* _TIFFFindFieldInfo(TIFF*, ttag_t, TIFFDataType); tif_dir.h:externconst TIFFFieldInfo* _TIFFFieldWithTag(TIFF*, ttag_t); tiffio.h:} TIFFFieldInfo; tiffio.h:const TIFFFieldInfo *info; tiffio.h:extern void TIFFMergeFieldInfo(TIFF*, const TIFFFieldInfo[], int); tiffio.h:extern const TIFFFieldInfo* TIFFFindFieldInfo(TIFF*, ttag_t, TIFFDataType); tiffio.h:extern const TIFFFieldInfo* TIFFFieldWithTag(TIFF*, ttag_t); tiffiop.h: TIFFFieldInfo** tif_fieldinfo; /* sorted table of registered tags */ what do u suggest me? -- Maurizio Marini ___ 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] dial '0' for outside line and get a dialtone...
On Friday 17 September 2004 11:43, Evert Meulie wrote: How do I implement this in extensions.conf...? maybe this may help... http://lists.digium.com/pipermail/asterisk-users/2004-February/036737.html -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 ___ 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] Re: hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.
On Saturday 11 September 2004 10:09, Hartmut Wahl wrote: Hello, i have exactly the same problem. Aug 24 04:47:57 weblogin kernel: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. I am running the card in NT-Mode, it happens some hours after zaphfc and asterisk are loaded. It takes a lot of CPU load for syslogging (the message repeats a lot) and asterisk connections are disturbed (fast my machine did hangup as growing logs fullfilled partition crackling). Last time it happend it was solved by unloading asterisk and zaphfc and loading it again. The first time I had it I had to reboot, otherwise ist startet just when I startet ztcfg. the messages repeat many times per second, syslog doesn't interval them with familiar: last message repeats times who knows? should i add MARCH=i586 option to asterisk/Makefile ? Did you try it, did it help? Any other hints? it does apply to asterisk, not to zaphfc :( it was a misleading suggestion, so i solved it installing in an other more powerful machine: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping: 10 cpu MHz : 999.556 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 1957.88 with this hw i've no issues at all; even strange messages i complained about in my previous posts like: PRI: !! Got a UA, but i'm in state 1 PRI: Double assgined TEI! disappeared by yestarday when i changed: ;pridialplan=national pridialplan=local IMHO, the pci bus performance cannot be below a threshold, i cannot imagine which could be it i hope this may help you Maurizio ___ 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] pridialplan nationalprefix
For whom which may be interested: Here in Italy we have GSM #numbers without leading zero PSTN instead has prefix starting with '0' to have '0' recognized by * i need to insert nationalprefix=0 as Jason Williams suggested me in irc; now, you cannot have: pridialplan=natonal otherwise * will not be able to call GSM phones you need to setup: pridialplan=local prilocaldialplan=local nationalprefix=0 Maurizio ___ 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] zaphfc errors
bri-stuff.0.1.0-RC4a on debian sid with 2.6.7 kernel isdn sk: Bus 0, device 18, function 0: Network controller: Digi International Datafire Micro V (Europe) (rev 2). IRQ 17. Master Capable. Latency=16. Max Lat=16. I/O at 0xe400 [0xe407]. Non-prefetchable 32 bit memory at 0xd8021000 [0xd80210ff]. i still receive error messages continuosly (every few seconds) i get in asterisk.log: Sep 9 10:14:33 WARNING[1105357744]: No D-channels available! Using Primary on channel anyway 3! Sep 9 10:14:34 WARNING[1105357744]: PRI: !! Got a UA, but i'm in state 1 Sep 9 10:15:01 WARNING[1105357744]: PRI: Double assgined TEI! Sep 9 10:15:21 DEBUG[1097968560]: Immediately destroying 1, having received INVAL at CLI i receive: received TEI check request for TEI = 127 Sep 9 10:58:17 WARNING[1105357744]: chan_zap.c:6902 zt_pri_error: PRI: Double assgined TEI! is there something wrong in italy isdn configuration? something that zaphfc does not reconize? apart from these errors, the system seems have no issues with PSTN Maurizio ___ 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] zaphfc strange errors
Hi i've an hfc-s card with last bristuff installed at cli i'm receiving: Sep 8 12:35:20 WARNING[1109552048]: chan_zap.c:6902 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 received TEI check request for TEI = 77 what is causing them? 10x Maurizio ___ 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] isdn, pbx and *
i have a traditional pbx attached to one line of NT1, and asterisk with hfc-s to the other one; when a call comes in, it is like asterisk captures it, passing it to the channel configured on dialplan; in the facts, the call is not answered, but NT1 doesn't ring pbx, as it would do if call had been answered. What is happening? Can i avoid this and have pbx and asterisk ringing simultaneously? 10x, Maurizio ___ 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] newbie question about PBX Call Pickup
Hi, sorry for annoying question; i read http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup without understanding: 1. how to add an ext. to a pickup group (ie:. how to populate pickup group) 2. how 'Directed pickup' does work? You dial the pickup number and your extension, and the call will only transfer if it is your extension should i digit something like '*8, then dial my extension? i tried to dial my extension but i got a busy tone maurizio ___ 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] hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, debian sid kernel 2.6.7 cpu: AMD Duron(tm) Processor kernel.log: Aug 23 17:33:40 weblogin kernel: Zapata Telephony Interface Registered on major 196 Aug 23 17:33:40 weblogin kernel: zaphfc: no version for zt_receive found: kernel tainted. Aug 23 17:33:40 weblogin kernel: PCI: Found IRQ 10 for device :00:0a.0 Aug 23 17:33:40 weblogin kernel: PCI: Sharing IRQ 10 with :00:07.5 Aug 23 17:33:40 weblogin kernel: zaphfc: Digi International Digi DataFire Micro V (Europe) configured at mem 0xe091a000 fifo 0xcf598000(0xf598000) IRQ 10 HZ 1000 Aug 23 17:33:40 weblogin kernel: zaphfc: Card 0 configured for TE mode Aug 23 17:33:40 weblogin kernel: zaphfc: 1 hfc-pci card(s) in this box. Aug 23 17:33:40 weblogin kernel: Registered tone zone 11 (Italy) Aug 24 04:47:57 weblogin kernel: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. should i add MARCH=i586 option to asterisk/Makefile ? any help will be very apreciated Maurizio - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBKyzZ4Q/49nIJTlwRAjiiAJ9UDhQOtnYBHgGlWzFW1Red12siIACfQgDA 7v66QWgn3FpFbPopHngLQ40= =g8pe -END PGP SIGNATURE- ___ 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] AgentLogin issue
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 13 August 2004 05:46, Joe Dennick wrote: Please note that the proper syntax is 'agent = agent-id, password, agent-name. i solved *all* my issue changing client (firefly in my case) to use iax2 instead of sip; agent 1001/1001 logins in, answer calls, etc etc now, why wit iax2 i have no problem, and with sip it doesn't work? incredible but true :) m. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBHIE84Q/49nIJTlwRAqelAJ9EXR7BHK4w7uI7jjZPk/WbaR5AEACfRG9T rW2hdodrZlGWQSj2pj1zZcw= =Pl9Y -END PGP SIGNATURE- ___ 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] AgentLogin issue
Hi
i have an issue getting agentLogin working
/etc/asterisk/queues.conf
member = Agent/1001
member = Agent/1002
extension.conf
exten = 110,1,Wait,1
exten = 110,2,AgentLogin()
now, i call 110 by a firefly client, trying to login in as 1001 agent:
Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact
hop: sip:[EMAIL PROTECTED]:5060
-- Executing Wait(SIP/sip3-768a, 1) in new stack
-- Executing AgentLogin(SIP/sip3-768a, ) in new stack
Aug 12 16:31:37 DEBUG[1127562160]: rtp.c:1156 ast_rtp_write: Ooh, format changed from
UNKN to ULAW
Aug 12 16:31:37 DEBUG[1127562160]: channel.c:1101 ast_settimeout: Scheduling timer at
160 sample intervals
-- Playing 'agent-user' (language 'en')
Aug 12 16:31:37 DEBUG[1103408048]: chan_sip.c:817 __sip_ack: Stopping retransmission
on '78383678327d335d' of Response 2: Found
Aug 12 16:31:41 DEBUG[1127562160]: channel.c:1101 ast_settimeout: Scheduling timer at
0 sample intervals
Aug 12 16:31:41 DEBUG[1127562160]: channel.c:1101 ast_settimeout: Scheduling timer at
0 sample intervals
Aug 12 16:31:42 DEBUG[1127562160]: rtp.c:189 send_dtmf: Sending dtmf: 49 (1), at
192.168.1.151
Aug 12 16:31:43 DEBUG[1127562160]: rtp.c:189 send_dtmf: Sending dtmf: 48 (0), at
192.168.1.151
Aug 12 16:31:44 DEBUG[1127562160]: rtp.c:189 send_dtmf: Sending dtmf: 48 (0), at
192.168.1.151
Aug 12 16:31:46 DEBUG[1127562160]: rtp.c:189 send_dtmf: Sending dtmf: 49 (1), at
192.168.1.151
Aug 12 16:31:47 DEBUG[1127562160]: rtp.c:189 send_dtmf: Sending dtmf: 35 (#), at
192.168.1.151
== Spawn extension (local, 110, 2) exited non-zero on 'SIP/sip3-768a'
Aug 12 16:31:51 DEBUG[1127562160]: cdr_addon_mysql.c:178 mysql_log: cdr_mysql:
inserting a CDR record.
Aug 12 16:31:51 DEBUG[1127562160]: cdr_addon_mysql.c:197 mysql_log: cdr_mysql: SQL
command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield)
VALUES ('2004-08-12 16:31:36','\sip3\ 103','103','110','local',
'SIP/sip3-768a','','AgentLogin','',15,14,'ANSWERED',3,'','1092321096.2','')
my call is interpreted as a phon call and cdr record it :(
what am i missing?
thnx for help
m.
--
Maurizio Marini GSM +39-335-8259739
Work: +39-0721-855285 Fax +39-0721-859609
Home: +39-0721-950396 IAXTel: (700) 350-1234
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Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI
On Monday 09 August 2004 14:06, Nick Barnes wrote: Holger Schurig: Basically yes, but ... Many thanks for your help - I'll stop playing with the AVM cards now! but chan_capi is in sync with * cvs, hfc-s support (bri_stuff) no -- Maurizio Marini ___ 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] avm c4
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi there, now c4 does work :) i plugged isdn cable in the fourth controller instead of the first one; now, the problem is: why the 4th does work and the 1th does not? i will try the 2th and 3th in the morning 10x - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBEJU94Q/49nIJTlwRAjDWAJ9yHX72cUhA0txJg6G6DtgwbM8o4QCdGSIu GazjEpC45MpcUcoh3JT4kug= =VGAm -END PGP SIGNATURE- ___ 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] capturing a call
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ddoes it feasible with * to capture a call? when arrives a call, floor bells ring and everyone can hear them in the company, then everyone can answer 'capturing' the call m. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBEJYq4Q/49nIJTlwRAsZtAJ9jDbfeLg9ia2n3yYy6RR3NBidY/wCcDgML O3ViqrM+ypEzAra3UOfZTVM= =0csq -END PGP SIGNATURE- ___ 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] avm c4, ptmp
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 02 August 2004 18:39, Deti Fliegl wrote: Your Extension has to match your MSNs. You have to configure all MSNs you have in a comma separated list like msn=27849,27852,27869,27861 and you must only use these MSNs as caller id. Hi :) thnx for having tryied to help :) we have 2 number on our isdn: 0721855285 and 0721859609 i try to call my home: 0721950396 here the issue: now in capi.conf i've: # cat capi.conf ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] [controller1] msn=0721855285,0721859609 incomingmsn=* controller=1,2,3,4 softdtmf=1 accountcode= context=default ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 mode=immediate isdnmode=p2mp ; ;-- in extension.conf i have: [local] ignorepat = 9 exten = _9XX.,1,Dial,CAPI/0721855285:bBYEXTENSION:1 exten = _9XX.,2,Congestion exten = _9XX.,3,Hangup Aug 3 11:26:31 DEBUG[1145346992]: chan_sip.c:4423 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 -- Executing Dial(SIP/sip1-5fcd, CAPI/0721855285:bBYEXTENSION:1) in new stack -- data = 0721855285:b90721950396:1 -- capi request omsn = 0721855285 Aug 3 11:26:31 NOTICE[1224625072]: chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn = 0721855285. you should check your config! Aug 3 11:26:31 NOTICE[1224625072]: app_dial.c:714 dial_exec: Unable to create channel of type 'CAPI' as yuo can see, -- data = 0721855285:b90721950396:1 -- capi request omsn = 0721855285 everithing seems ok :) byez Maurizio - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBD12N4Q/49nIJTlwRApRtAJ94VfuG+F00IJRuyIz7vbajjLOmggCfcAwT RFhrkbzXB3TBqcieHz5k74A= =Pms4 -END PGP SIGNATURE- ___ 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] avm c4, ptmp
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jason :) On Tuesday 03 August 2004 12:07, Jason Williams wrote: I would set the MSN's to 855285 and 859609 They do not usually include the area code. [local] exten = _9XX.,1,Dial,CAPI/855285:bBYEXTENSION:1 exten = _9XX.,2,Congestion exten = _9XX.,3,Hangup ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] [controller1] msn=855285,859609 incomingmsn=* controller=1,2,3,4 softdtmf=1 accountcode= context=local ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 mode=immediate isdnmode=p2mp ; ;-- Aug 3 12:02:28 DEBUG[1145346992]: chan_sip.c:4423 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 -- Executing Dial(SIP/sip1-0167, CAPI/855285:bBYEXTENSION:1) in new stack -- data = 855285:b90721950396:1 -- capi request omsn = 855285 Aug 3 12:02:28 NOTICE[1224625072]: chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn = 855285. you should check your config! Aug 3 12:02:28 NOTICE[1224625072]: app_dial.c:714 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy/congested at this time - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBD2W24Q/49nIJTlwRAi0cAJ4/ckdwqJMDbWVYYsMU8wj9zksbugCeJfl5 lh2CHTrKNg7WOhqfFf/B1Zo= =LVNs -END PGP SIGNATURE- ___ 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] avm c4: DISCONNECT_IND ID=001 #0x0193 LEN=0014
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 i fixed wrong capi.conf (there was [controller1] after [interfaces]) now capi.conf is: ; ; CAPI config ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=855285,859609 incomingmsn=* controller=1,2,3,4 softdtmf=0 accountcode= context=local ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=4 mode=immediate isdnmode=ptm question: devices should be 4 or 2? now when a issue a call i get: Aug 3 14:43:45 DEBUG[1224625072]: pbx.c:1255 pbx_extension_helper: Launching 'Dial' -- data = 855285:0721950396 -- capi request omsn = 855285 == found capi with omsn = 855285 Urgent handler == CAPI Call CAPI[contr1/855285]/6 -- Called 855285:0721950396 Urgent handler -- CONNECT_CONF ID=001 #0x0010 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 -- CONNECT_CONF ID=001 #0x0010 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 Urgent handler Aug 3 14:43:45 DEBUG[1224625072]: channel.c:1699 ast_set_read_format: Set channel CAPI[contr1/855285]/6 to read format ULAW Aug 3 14:43:45 DEBUG[1224625072]: channel.c:1666 ast_set_write_format: Set channel SIP/sip1-9316 to write format ULAW Aug 3 14:43:45 DEBUG[1224625072]: channel.c:1666 ast_set_write_format: Set channel CAPI[contr1/855285]/6 to write format ALAW Aug 3 14:43:45 DEBUG[1224625072]: channel.c:1699 ast_set_read_format: Set channel SIP/sip1-9316 to read format ALAW -- DISCONNECT_IND ID=001 #0x0193 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3301 == DISCONNECT_IND PLCI=0x101 REASON=0x3301 Urgent handler someone knows? 10x - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBD40e4Q/49nIJTlwRAnq/AJ0dJ3ybyYOlh8xtQYDdvS4xT3BNLwCeN74p r7OJfCwcpDqccyKq1S+YWXA= =pfSZ -END PGP SIGNATURE- ___ 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] avm c4, ptmp
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi there, i'm in debian sid 3.1 with kernel 2.6.7, * last cvs chan_capi 0.3.4b; nt1+ with 2 bri in ptmp (http://www.voip-info.org/tiki-index.php?page=DDI) i tried to install avm c4 following step by step http://www.voip-info.org/tiki-index.php?page=Asterisk%20How%20to%20connect%20with%20CAPI step 1. i compiled capi 2.0 support in kernel 2.6.7 step 2. modprobe c4 to insmoded kernel module then: You may need firmware in /usr/lib/isdn/ and a suitably configured /etc/capi.conf i googled around for a suitable firmware (c4.bin) but the only one i was able to find is this: ftp://ftp.in-berlin.de/pub/capi4linux/firmware/c4/3-11-04 i cannot say if this one is ok or something other should be found installed then i apt-got capi utils to have capiinit and capiinfo; you need capiinit to install firmware; using my knwoledge, the only way to have /usr/lib/isdn/c4.bin is installling suse 9, but this is not my last wish at teh moment this is my /etc/isdn/capi.conf: c4 /usr/sbin/c4.binDSS1- - - - capiinit now start c4 with no complain; in /etc/asterisk/capi.conf i have: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] [controller1] msn=0xx incomingmsn=* controller=1 softdtmf=1 accountcode= context=default ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 ;mode=immediate isdnmode=p2mp asterisk starts without any complain but i'm unable to send or receive calls; when i receive a call never show up in CLI; when i issue an outside call i get: -- Executing Dial(SIP/sip1-07f4, CAPI/0721xx:bBYEXTENSION:1) in new stack -- data = 0721xx:b0721950396:1 -- capi request omsn = 0721xx Aug 2 17:53:02 NOTICE[1224547248]: chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn = 0721xx. you should check your config! any help is apreciated m. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBDmZk4Q/49nIJTlwRAh0eAJ44pF18ws1pf4M6iUcqzTxJx8LM/gCfSC+Y o22/axRVPCei6GLGCaJUpMA= =1+La -END PGP SIGNATURE- ___ 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] Softphone - Freeware?!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 You do need to have this enabled in the dialplan dial strings to enable transfers. u should use something like this: [from-sip] exten = 101,1,Dial(SIP/sip1,20,tTr) from http://www.voip-info.org/wiki-Asterisk+cmd+Dial: The options parameter, which is optional, is a string containging zero or more of the following flags and paramters: t: Allow the called user to transfer the call T: Allow the calling user to transfer the call r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBClf24Q/49nIJTlwRAg/kAJ90/tEQZXEIVe+A1WTM7HDtQGF1dgCeLrCG 01E4lkdvIbjpGrvMoiGu324= =lgdl -END PGP SIGNATURE- ___ 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] chan_capi-0.3.4b and asterisk last cvs
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi i've installed asterisk by last cvs and i note res_parking.c is not anymore there; chan_capi-0.3.4b INSTALL file require: in /etc/asterisk/modules.conf insert the line: load = res_parking.so load = chan_capi.so running asterisk i get: [app_capiCD.so]Jul 21 15:32:26 WARNING[1076988448]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber Jul 21 15:32:26 WARNING[1076988448]: loader.c:423 load_modules: Loading module app_capiCD.so failed! how can i fix the issue? 10x for help - -- Maurizio Marini -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA/nL14Q/49nIJTlwRAgJWAJ98lB9iOAODqf8jyYodchA+DyGhjACfb2ET vkA7cpMw2qa89jQF2vtCeaY= =15Ly -END PGP SIGNATURE- ___ 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] HFC-S card and Unable to create channel of type 'Zap'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 hi, i'm new to * I've installed an hfc-s card (DIGI Micro V) with bristuff 0.0.2; when i try to call outside i get: -- Accepting AUTHENTICATED call from 192.168.1.110, requested format = 1024, actual format = 1024 -- Executing Dial([EMAIL PROTECTED]/2, Zap/g1/0123456) in new stack Jul 13 13:42:49 NOTICE[884752]: app_dial.c:559 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time Jul 13 13:43:07 WARNING[163851]: chan_zap.c:6070 zt_pri_error: PRI: Read on 19 failed: Unknown error 500 Jul 13 13:43:07 NOTICE[163851]: chan_zap.c:6976 pri_dchannel: PRI got event: 6 on span 1 - /etc/zaptel.conf loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 - ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) 3 channels configured. - /etc/asterisk/zapata.conf [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode signalling = bri_cpe_ptmp pridialplan = local prilocaldialplan = local echocancel=yes immediate=yes group = 1 context = local channel = 1-2 *CLI zap show channel 1 Channel: 1 File Descriptor: 17 Span: 1 Extension: Context: local Caller ID string: Destroy: 0 Signalling Type: PRI Signalling Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7351 zap_show_channel: Failed to get conference info on channel 1 Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7357 zap_show_channel: Failed to get confmute info on channel 1 any help will be very apreciated 10x Maurizio - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA89R44Q/49nIJTlwRAtzBAJ9TPn4Hn6WKECiXYFr9Jnf3f0WrnwCePDX+ O1t5ts8wdlOzBU/HyqQpqh4= =Ujbh -END PGP SIGNATURE- ___ 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] HFC-S card and Unable to create channel of type 'Zap'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 13 July 2004 14:41, Alessio Focardi wrote: Ciao ! are you connecting a phone or a pbcx to the isdn card ? simply, i'm connecting this isdn card to an nt1 plus to call outside... - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA898z4Q/49nIJTlwRAtjvAJ4graOK+ODpNyBmrvQiisKF5CVF3wCfbMJR jhsOV93kXX5p8Ygm1SgJDNY= =0oqd -END PGP SIGNATURE- ___ 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
