RE: [Asterisk-Users] SIP NAT + m0n0wall 1:1 mapping
In my instance I'm using m0n0wall, but this is a hardware-neutral question. Sometimes, yes and no. The trick in Monowall I founds is to use the auto add in Monowall to create the rules. If you manually create the rule, she don' work. Why? Dunno. Using 1.11 built on Thu Nov 11 23:02:41 CET 2004 generic-pc-cdrom. Works behind my mother in law's crappy SMC router with appropriate rules created. Firewall NAT Interface Wan External Address External IP Protocol: tcp/udp (snom for example can pass either) Source: Any Port: Any / Any NAT IP Your Asterisk Server IP Local Port (Other) 5060 Description: Asterisk SIP rule Auto-add a firewall rule to permit traffic through this NAT rule is CHECKED! Firewall NAT Interface WAN External Address External IP Protocol tcp/udp Source 1-2 Port Any / Any NAT IP Your Asterisk Server IP Local port 1:2 Description: Asterisk Media Stream Rule Auto-add a firewall rule to permit traffic through this NAT rule is CHECKED! Tested w/ X-Ten Snom 190 YMMV HTH -Original Message- From: Robert Goodyear [mailto:[EMAIL PROTECTED] Sent: Monday, July 11, 2005 9:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP NAT + m0n0wall 1:1 mapping I know a SIP client behind a NAT trying to peer with Asterisk behind another NAT is troublesome. Has anyone had any luck doing this by interfacing Asterisk to the WAN using 1:1 NAT translation to give it a public IP while still firewalled? In my instance I'm using m0n0wall, but this is a hardware-neutral question. Thanks. -- Robert Goodyear Brand Up LLC http://www.brand-up.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 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] Modem Connection from TDM card to TE4xxP card
I just needed to test a dialup modem connection (don't ask) and I had a modem connected to a TDM card (FXS port) which then dialled out via a E1 PRI on a TE4xxp card. See my log below: atdt0198xx CONNECT 36000 V42bis ** Dial IP ** Username: Password: Entering PPP Session. IP address is xxx.xxx.xxx.xxx MTU is 1500. OK ATH0 OK ATDT0198xx CONNECT 31200 V42bis ** Dial IP ** Username: Password: Entering PPP Session. IP address is xxx.xxx.xxx.xxx MTU is 1500. OK ATh0 OK So it looks like there are at least some cases where the TDM cards do work very well even for modem connections. Some details of this setup: aster0x root # lspci :00:00.0 Host bridge: Intel Corp. 82875P Memory Controller Hub (rev 02) :00:03.0 PCI bridge: Intel Corp. 82875P Processor to PCI to CSA Bridge (rev 02) :00:1c.0 PCI bridge: Intel Corp. 6300ESB 64-bit PCI-X Bridge (rev 02) :00:1d.0 USB Controller: Intel Corp. 6300ESB USB Universal Host Controller (rev 02) :00:1d.1 USB Controller: Intel Corp. 6300ESB USB Universal Host Controller (rev 02) :00:1d.4 System peripheral: Intel Corp. 6300ESB Watchdog Timer (rev 02) :00:1d.5 PIC: Intel Corp. 6300ESB I/O Advanced Programmable Interrupt Controller (rev 02) :00:1e.0 PCI bridge: Intel Corp. 82801 PCI Bridge (rev 0a) :00:1f.0 ISA bridge: Intel Corp. 6300ESB LPC Interface Controller (rev 02) :00:1f.1 IDE interface: Intel Corp. 6300ESB PATA Storage Controller (rev 02) :00:1f.2 IDE interface: Intel Corp. 6300ESB SATA Storage Controller (rev 02) :00:1f.3 SMBus: Intel Corp. 6300ESB SMBus Controller (rev 02) :01:01.0 Ethernet controller: Intel Corp. 82547GI Gigabit Ethernet Controller :02:02.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) :02:04.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface :03:00.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) :03:01.0 Ethernet controller: Intel Corp. 82557/8/9 [Ethernet Pro 100] (rev 10) aster0x root # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 3 model name : Intel(R) Pentium(R) 4 CPU 3.20GHz stepping: 4 cpu MHz : 3193.205 cache size : 1024 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe pni monitor ds_cpl cid bogomips: 6307.84 aster0x root # cat /proc/interrupts CPU0 0: 1598414593IO-APIC-edge timer 1: 1540IO-APIC-edge i8042 8: 4IO-APIC-edge rtc 10: 0 IO-APIC-level acpi 12:842IO-APIC-edge i8042 14: 1IO-APIC-edge ide0 17:8599800 IO-APIC-level eth1 18: 62398983 IO-APIC-level libata, eth0 24: 1598214277 IO-APIC-level t4xxp 26: 1598215786 IO-APIC-level wctdm - BTW, module is actually wcfxs NMI: 0 LOC: 1598524578 ERR: 0 MIS: 0 Asterisk version, and likely zaptel/libpri are from the same date: aster0x asterisk # cat .version CVS-v1-0-06/02/05-18:22:29 aster0x zaptel # ./zttest Opened pseudo zap interface, measuring accuracy... 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% 100.00% 100.00% 100.00% 99.987793% --- Results after 63 passes --- Best: 100.00 -- Worst: 99.987793 Feel free to request additional information or tests... Regards, Adam ___ 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] Zaptel won't compile under Fedora Core 4
On 7/11/05, Gonzalo Servat [EMAIL PROTECTED] wrote: On 7/12/05, Eric Bullen [EMAIL PROTECTED] wrote: I hope someone can offer me some help with this. Basically, the current CVS version of Zaptel will not compile under Fedora Core 4. I have closely followed the directions in http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 using the versions given in the FC4 distro with no luck.Here's the output when I run make linux26. Any help would be great. TIA.[...snip...]In file included from /asterisk_source/zaptel/zaptel.c:40:/asterisk_source/zaptel/zconfig.h:10:27: error: linux/version.h: No such file or directoryTry installing kernel-source and/or glibc-kernheaders RPMs from the FC4 CDs. I did this already- it's outlined in the URL I listed above. Any other suggestions? ___ 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] Zaptel won't compile under Fedora Core 4
On Tue, Jul 12, 2005 at 11:04:16AM +1000, Gonzalo Servat wrote:
On 7/12/05, Eric Bullen [EMAIL PROTECTED] wrote:
I hope someone can offer me some help with this. Basically, the current CVS
version of Zaptel will not compile under Fedora Core 4. I have closely
followed the directions in
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
using the versions given in the FC4 distro with no luck. Here's the output
when I run make linux26. Any help would be great. TIA.
[...snip...]
In file included from /asterisk_source/zaptel/zaptel.c:40:
/asterisk_source/zaptel/zconfig.h:10:27: error:
linux/version.h: No such file or directory
Try installing kernel-source and/or glibc-kernheaders RPMs from the FC4 CDs.
kernel-glibcheaders is the part of the kernel headers that user-space
programs need. Not any good reference for any kernel module to build
with.
linux/version.h is generated as part of the configuration process of the
kernel configuration process (make {,menu,x,g}config) of the kernel
source.
What version of zaptel do you try to build? For what kernel version?
Your distro's default or your one you've built yourself?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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RE: [Asterisk-Users] Pushing new firmware to Snom 190
Take a look at http://www.snom.com/white_papers.html, http://www.snom.com/whitepapers/FAQ-04-03-26-v3_4-sf.pdf and check out DHCP option 66 and 67. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Monday, July 11, 2005 11:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Pushing new firmware to Snom 190 Anyone know how I can push a firmware update to a Snom 190 without using DHCP? In the web interface, I specify a path to the Snom firmware, and it works, except I have to physically press OK to get the update to download. I need to do it remotely... ___ 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 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] IP Phone with Standard Power Ethernet
After IEEE finally released 802.3af snom supports all three modes in the 320/360 models: http://www.snom.com/whitepapers/faq-05-03-16-da.pdf (snom 320 = snom 360). CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chris gamble Sent: Tuesday, July 12, 2005 5:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP Phone with Standard Power Ethernet I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip phone that uses IEEE power over ethernet -- or is there a problem with IEEE power over ethernet?? ___ 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 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] IP Phone with Standard Power Ethernet
The new grandstream GXP-2000 works quite nice with the standard 802.3af David Christian Stredicke wrote: After IEEE finally released 802.3af snom supports all three modes in the 320/360 models: http://www.snom.com/whitepapers/faq-05-03-16-da.pdf (snom 320 = snom 360). CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of chris gamble Sent: Tuesday, July 12, 2005 5:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IP Phone with Standard Power Ethernet I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip phone that uses IEEE power over ethernet -- or is there a problem with IEEE power over ethernet?? ___ 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 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 -- - David Liu Chief Operating Officer Deltapath Commerce Technology Limited HK Tel: +852 3107-1333 HK Cell: +852 9166-1880 US Tel: +1 313 228-0906 - SIP Technology Provider! www.deltapath.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
Re: [Asterisk-Users] Cisco 7960 firmware upgrade promblems
Marc Fishman ha scritto: I appreciate the response but that's what isn't working. I have tried v5.3 and v3.0 with the same result. I suspect the firmware version (P003AM30) is I know it's hard to find out infos at the cisco site. Maybe you can open a TAC case Sergio ___ 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] Asteriski misses the table
I am not aware what I have done wrong, but the result is a query of:
*Database error:* Invalid SQL: SELECT * FROM WHERE
UNIX_TIMESTAMP(calldate) = UNIX_TIMESTAMP('2005-07-01') ORDER BY
calldate DESC LIMIT 0,25
*MySQL Error*: 1064 (You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right
syntax to use near 'WHERE UNIX_TIMESTAMP(calldate) =
UNIX_TIMESTAMP('2005-07-01') ORDER BY calldat' at line 1)
*Database error:* Invalid SQL: SELECT count(*) FROM WHERE
UNIX_TIMESTAMP(calldate) = UNIX_TIMESTAMP('2005-07-01')
*MySQL Error*: 1064 (You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right
syntax to use near 'WHERE UNIX_TIMESTAMP(calldate) =
UNIX_TIMESTAMP('2005-07-01')' at line 1)
*Database error:* next_record called with no query pending.
*MySQL Error*: 1064 (You have an error in your SQL syntax; check the
manual that corresponds to your MySQL server version for the right
syntax to use near 'WHERE UNIX_TIMESTAMP(calldate) =
UNIX_TIMESTAMP('2005-07-01')' at line 1)
It misses the table.
I have set the table name in the /lib/defines.php
(I checked several times for a spelling error, quotemark, ...)
bye
Ronald
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[Asterisk-Users] Asterisk realtime failover problems
Dear All, I was trying to use Realtime Asterisk option for Sip users and peers + Heartbeat + Mysql Replication in order to make a failover system, so that if Ast1 went down for any reason, Ast2 server will have the same data that Ast1 used in the Mysql database and don't need to make the phones re-register. But when I started testing: the calls that where active during the transition between the two servers where disconnected (the two phones are talking peer to peer thanks to the canreinvite option but they we still sending UDP packets to port 1025 to the asterisk server), the phones must re-register with the new server though the Mysql server was replicated and the new server should have the data it needs. Has anyone trid doing this before, or does anyone have any idea if this should work or is there another way to do so, I will really appreciate it very much if anyone has any helping pointers. -- Thx MAG ___ 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] chan_capi ASTCC trouble
On 10 Jul 2005 at 22:01, Armin Schindler wrote: On Sun, 10 Jul 2005, Clive wrote: Hi all I am wondering if anyone has had a similar trouble to this: The timeout arguments in the dial command does not work. The caller does not get disconnected when the timeout reaches zero. I am not sure if this is a chan_capi issue, or a asterisk issue. I am using CVS-head and chan_capi CVS head also. Any suggestions or help will be appreciated. Thanks Clive Ok, just did some testing on the dial command using only iax2 and it does disconnect the call, so this may be a chan_capi issue. As far as I know, the timeout and hangup logic is done within Asterisk e.g. dial-application. chan-capi does not know anything about a timeout, so I don't know how this can be the location of the problem. Armin Hi On doing some tests, I have found that the timeout works fine only if the caller does not dial any DTMF tones , like for an IVR system. If DTMF tones are dialled during the call, the timeout doesn't work. another piece to add to the puzzle..:) very wacky, but hopefully this may help find the bug best regards Clive ___ 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] IP Phone with Standard Power Ethernet
exactly, only high-end cisco 7970/71 are 802.3af compliant, other models (7905-7960) using proprietary PoE detection and you will be out of luck if you use non-ci$co poe equipment, as I know, powerdsine midspans (because have cisco detection support) can power cisco 7912 directly (without pasive polarity dongle), 7940/60 can be powered only using this dongle and with powerdsine... you are totaly out of luck with cisco phones and e.g. Planet poe midspans (using standard 802.3af detection) I tried this some months ago, so this is from practice. PJ Kevin P. Fleming wrote: Tom wrote: Cisco supports 802.3af with a special cross-over cable which is easy and cheap to make. Nope, Cisco phones do _not_ support 802.3af. With the cross-over cable you can make them work off a 'dumb' power injector, but a true 802.3af injector without Cisco-proprietary support will not supply them power, as they will not negotiate for it since they don't support the same protocol. On top of that, a switch with built-in PoE can choose to supply power on the _data_ wires instead of the spare wires (which is fully supported by the spec), in which case a Cisco phone would have no hope of working. ___ 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 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 not accepting user input .. pls help !!
Hi guys,
I currently have a sip proxy server (sip express router) which
registers the sip phones. I need to add voice mail capability, i.e.
sip express router will forward all incoming calls to Asterisk if the
user does not pick up the call in 15 seconds.
The voicemail recording stops when the user hangs up. However, the
recording does not end if the user presses the # key, i.e. it is
ignoring the user input.
Similarly, when the user dials 2102 to access his voice mail, Asterisk
plays the prompt, but it seems to ignore all the user input keys.
Please kindly advise.
Regards,
YY
*
Config files
--
1) Ser
-
ser.cfg (SER)
-
# -- tm params --
# set time for which ser will be waiting for a final response;
# fr_inv_timer sets value for INVITE transactions,
# fr_timer for all others
modparam(tm,fr_inv_timer,15)
modparam(tm,fr_timer,10)
if (uri==myself) {
if (method==REGISTER) {
# attempt handoff to PSTN
if (uri=~^sip:[EMAIL PROTECTED]) {## This assumes
that the caller
log(1, Forwarding to PSTN\n); ## is
registered in our realm
forward(10.10.10.3, 5060); ## Our
Cisco router
break;
};
# retrieve voicemail
#
if (uri=~^sip:[EMAIL PROTECTED]) {
log(1, Retrieving voicemail\n);
# redirect now!
rewritehostport(202.125.25.102:5061);
append_branch();
t_relay_to_udp(202.125.25.106,5061);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup(location)) {
sl_send_reply(404, Not Found);
break;
};
timeout occurred ... now to forward to Asterisk's voicemail
service
if(method == INVITE)
{
t_on_failure(1);
};
};
t_relay();
# leave voicemail
#
failure_route[1] {
log(1,Activating voicemail!!\n);
revert_uri();
# redirect now to Asterisk (on the same machine) !
rewritehostport(202.125.25.102:5061);
append_branch();
t_relay_to_udp(202.125.25.106,5061);
}
2) Asterisk
sip.conf
[general]
context=test
port=5061 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; ip phone 1012, registered with SER
[1012]
type=friend
username=1012
canreinvite=no
context=test
mailbox=1012
host=203.125.25.106
nat=no
dtmfmode=info
disallow=all
allow=alaw
allow=ulaw
---
extensions.conf
-
[test]
;leave voice messages
exten = 1012,1,Voicemail(u1012)
exten = 1012,2,Hangup
;play voice messages
exten = 2012,1,VoiceMailMain,1012
exten = 2012,2,Hangup
-
voicemail.conf
[default]
1012 = 1234, YY, [EMAIL PROTECTED]
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[Asterisk-Users] Asterisk realtime failover problems
Dear All, I was trying to use Realtime Asterisk option for Sip users and peers + Heartbeat + Mysql Replication in order to make a failover system, so that if Ast1 went down for any reason, Ast2 server will have the same data that Ast1 used in the Mysql database and don't need to make the phones re-register. But when I started testing: the calls that where active during the transition between the two servers where disconnected (the two phones are talking peer to peer thanks to the canreinvite option but they we still sending UDP packets to port 1025 to the asterisk server), the phones must re-register with the new server though the Mysql server was replicated and the new server should have the data it needs. Has anyone trid doing this before, or does anyone have any idea if this should work or is there another way to do so, I will really appreciate it very much if anyone has any helping pointers. -- Thx MAG ___ 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] monitor using incorrect path
Hello, I have been noticing the following behaviour with the monitor command.. Normally it records to the default location and then uses soxmix to create the correct wav file. But for some reason sometimes it doesn't use /var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/.. (notice the 2 // in front!) Here is some logging: monitor executing ( nice -n 19 soxmix //var/spool/asterisk/monitor/SIP-242-027e_1-in.wav //var/spool/asterisk/monitor/SIP-242-027e_1-out.wav //var/spool/asterisk/monitor/SIP-242-027e_1.wav rm -f //var/spool/asterisk/monitor/SIP-242-027e_1-* ) And when it is correct, it does: monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/SIP-220-c400_0-in.wav /var/spool/asterisk/monitor/SIP-220-c400_0-out.wav /var/spool/asterisk/monitor/SIP-220-c400_0.wav rm -f /var/spool/asterisk/monitor/SIP-220-c400_0-* ) For the record, I am using bristuff-RC8h (that is, quadBRI and asterisk-1.0.8) on a Debian 3.1. Any ideas on what I might be doing wrong, or does anyone see the same behaviour? Cheers, Kristof ___ 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] Re: DTMF not sending properly via IAX
In article [EMAIL PROTECTED], Mark Edwards [EMAIL PROTECTED] wrote: Hi Tony I am having a similar issue to you - from the 'other' direction in that when I connect to * via IAX2 the DTMF is being ignored. I am running HEAD at the moment. (and for the benefit of another subscriber so that they don't have to invoke their autoresponder I acknowlege that DTMF is sent out of band over IAX...) http://bugs.digium.com/view.php?id=4631 Does this bear any relation? I'm not sure. Incoming DTMF using IAX2 from the PSTN via my provider to my box seems to work fine. When my box tries to send DTMF, it appears that the first digit of the string is translated by the provider into DTMF tones over the PSTN, but then subsequent digits in the same string, although sent by IAX2 from my box, are not translated by the provider. I remember reading something some time ago about DTMF getting lost due to sequence number problems, but I can't remember whether that was SIP or IAX. I'm hoping to avoid having to delve into the code myself, due to lack of time, and I'm not even sure whether the problem is my end or the provider's. Cheers Tony Mark On 7/12/05, Tony Mountifield [EMAIL PROTECTED] wrote: I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have the following dialplan entries, that I can dial into: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,SendDTMF(1234567890) exten = s,4,Wait(2) exten = s,5,Hangup What happens is that whatever string I give to SendDTMF, I only hear the first DTMF digit. The remaining digits don't get sent. I recently updated from an April CVS-STABLE to the July 4 version, but I couldn't see any relevant differences in the code. As its a production system, I can't just revert to test without planning. I don't know whether Voiptalk have changed anything. Is this a known bug in certain versions of Asterisk? If I do iax2 debug, I *can* see DTMF frames being sent and acked for each digit. Wo I can't understand why I'm not hearing all the digits. -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] asking again
ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. _ FREE pop-up blocking with the new MSN Toolbar get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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 PBX and Siemens Hipath 3750
Hello I am planning to build a small PBX using TDM22B. We have a Siemens Hipath 3750 in operation already. When I manage to complete my PBX using TDM22B I would ofcourse like to be able to connect my Asterisks PBX with the Siemens Hipath 3750 PBX. Will there be any issues regarding my plan ? Or is there any other issues that I need to take into account vis-a-vis Siemens PBX. I have never done all this before so I would appreciate any inputs. Thanks in advance Varun ___ 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] Re: h323 and asterisk
In article [EMAIL PROTECTED],
Ronald_Wiplinger [EMAIL PROTECTED] wrote:
We come into this section of the dialplan:
exten = 8867033,1,Wait(1)
exten = 8867033,n,SayUnixTime
exten = 8867033,n,NoOp(If you know the extension ...)
exten = 8867033,n,Dial(${PHONE_6003})
The caller from the GK hears only ringing, not the time.
The extension 6003 rings and I can pick up, but without any voice nor video.
Try putting an Answer before the Wait.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Re: [Astrik-Usrs] callto: URL (URI) tag for dialing
Hello! Can you please post your CGI script ? Thanks M.G. - Original Message - From: Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 24, 2005 1:30 AM Subject: RE: [Asterisk-Users] callto: URL (URI) tag for dialing I just wrote a simple cgi to have a form generate the number, then the cgi creates a call file and bingo. Web call. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Elkins Sent: Friday, April 22, 2005 8:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] callto: URL (URI) tag for dialing I see that there seems to be a 'callto' URL/URI for dialling a phone number... ie - on my web site's Contact Page - I have added the code... a href=callto:+27128070590+27 12 807-0590/a There should be some generic way for Mozilla (firefox - etc) to somehow turn a click on such a link into persuading Asterisk to dial the number for me and connect it to my SIP hard-phone. 1 - mini application under mozilla to collect the number/sip address, add in a static local extension (personal settings?) and pass info to a listener (auto-dialer) on the Asterisk Machine 2 - Auto Dialer dials my extension, then on answer, dials the URL or phone number. The URL could either be a simple phone number or a full SIP address?? Anyone done this? ..and care to share? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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 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 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] bristuff patches and realtime mysql
Hi! I have a problem compiling the res_config_mysql.so after successfully compiling and installing asterisk with the bristuff package/patches. I get lots of compiler errors. When inserting a previously compiled res_config_mysql.so into the bristuff-patched Asterisk, I get an error on startup and Asterisk crashes. Anyone using res_config_mysql.so with the bristuff package and can help me? Thanks, Christoph ___ 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] monitor using incorrect path
On Tue, Jul 12, 2005 at 09:52:09AM +0200, Kristof Hardy wrote: Hello, I have been noticing the following behaviour with the monitor command.. Normally it records to the default location and then uses soxmix to create the correct wav file. But for some reason sometimes it doesn't use /var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/.. (notice the 2 // in front!) Here is some logging: monitor executing ( nice -n 19 soxmix //var/spool/asterisk/monitor/SIP-242-027e_1-in.wav //var/spool/asterisk/monitor/SIP-242-027e_1-out.wav //var/spool/asterisk/monitor/SIP-242-027e_1.wav rm -f //var/spool/asterisk/monitor/SIP-242-027e_1-* ) Logging from what exactly? That shouldn't be a problem on any posix system (except cygwin) . '//' is simply translated to '/' . I suspect you have a different problem. And when it is correct, it does: monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/SIP-220-c400_0-in.wav /var/spool/asterisk/monitor/SIP-220-c400_0-out.wav /var/spool/asterisk/monitor/SIP-220-c400_0.wav rm -f /var/spool/asterisk/monitor/SIP-220-c400_0-* ) For the record, I am using bristuff-RC8h (that is, quadBRI and asterisk-1.0.8) on a Debian 3.1. Any ideas on what I might be doing wrong, or does anyone see the same behaviour? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] chan_capi-cm-0.5.3 and ${DNID}
Hiya!
I have just moved up to asterisk 1.0.9, and also to chan_capi-cm-0.5.3.
Since the upgrade, the ${DNID} variable seems not to be set anymore.
I made the updates to modules.conf (and in the globals sections). When
in Debug, CALLERID is correct - but DNID is NULL.
any ideas?
Álainn
The cheese-mites asked how the cheese got there,
And warmly debated the matter;
The orthodox said that it came from the air,
And the heretics said from the platter. Anon.
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Re: [Asterisk-Users] asking again
Hi Wasim, Check out the x-lite softphone http://www.xten.com/ As for linux check this page there are two softphones type available : http://www.iptel.org/products Thx MAG wassim Darwish wrote: ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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 -- Thx MAG ___ 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] asking again
On Tue, Jul 12, 2005 at 08:31:07AM +, wassim Darwish wrote: ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. Depends on what you want from it. e.g: iaxcomm is free and availble for both those platforms. OTOH, the user interface, well, leaves some room for improvements. Have you actually followed recent discussions in the list about this subject? Have you checked the wiki? BTW: the wiki page is indeed quite confusing for someone who justs wants a softphone that works. Though I'm not sure exactly what are the questions people ask themselves when coming to that page and thus how to best answer them. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Last CVS - High Load
Good morning on our Test Machine based on RedHat 9 on a Pentium 4 SMP we are experiencing a 99% CPU load for asterisk with the CVS of tonight does anyone noticed that best regards Thierry ___ 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] monitor using incorrect path
Hi, I'm using Bristuff 0.2.0-RC7k and asterisk 1.0.6 and I'm facing something similar... Nearly all my monitor files are in 2 parts, soxmix doesn't compile them into one file. But I don't think soxmix is to blame because when I run it from the command line, everything is ok... the problem seems to originate from the command line executing soxmix. Help would be appreciate on that matter. Best regards David -Message d'origine- De : Tzafrir Cohen [mailto:[EMAIL PROTECTED] Envoyé : mardi 12 juillet 2005 10:45 À : asterisk-users@lists.digium.com Objet : Re: [Asterisk-Users] monitor using incorrect path On Tue, Jul 12, 2005 at 09:52:09AM +0200, Kristof Hardy wrote: Hello, I have been noticing the following behaviour with the monitor command.. Normally it records to the default location and then uses soxmix to create the correct wav file. But for some reason sometimes it doesn't use /var/spool/asterisk/monitor/.. but //var/spool/asterisk/monitor/.. (notice the 2 // in front!) Here is some logging: monitor executing ( nice -n 19 soxmix //var/spool/asterisk/monitor/SIP-242-027e_1-in.wav //var/spool/asterisk/monitor/SIP-242-027e_1-out.wav //var/spool/asterisk/monitor/SIP-242-027e_1.wav rm -f //var/spool/asterisk/monitor/SIP-242-027e_1-* ) Logging from what exactly? That shouldn't be a problem on any posix system (except cygwin) . '//' is simply translated to '/' . I suspect you have a different problem. And when it is correct, it does: monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/SIP-220-c400_0-in.wav /var/spool/asterisk/monitor/SIP-220-c400_0-out.wav /var/spool/asterisk/monitor/SIP-220-c400_0.wav rm -f /var/spool/asterisk/monitor/SIP-220-c400_0-* ) For the record, I am using bristuff-RC8h (that is, quadBRI and asterisk-1.0.8) on a Debian 3.1. Any ideas on what I might be doing wrong, or does anyone see the same behaviour? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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 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] asking again
Those 2 softphones below only do SIP, no IAX. Zoa, -- www.asteriskguru.com Mohamed A. Gombolaty wrote: Hi Wasim, Check out the x-lite softphone http://www.xten.com/ As for linux check this page there are two softphones type available : http://www.iptel.org/products Thx MAG wassim Darwish wrote: ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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 -- Thx MAG ___ 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 signature.asc Description: OpenPGP digital signature ___ 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] DTMF not sending properly via IAX
Not sure this applies, but a few months ago some of us were having problems with an itsp (livevoip.com) and incoming iax calls that hit the * ivr. dtmf was essentially not being passed. I opened a bug, however Mark quickly closed it with a note that's the way iax works; have the itsp fix their problem. The issue at that time was oriented around a call that was considered answered. If the end-to-end path of the call is considered answered, asterisk is no longer in the voice path to translate dtmf into iax out-of-band dtmf packets. Part of the argument was then what is an asterisk user suppose to do to pass dtmf across an iax link that uses g729? g729 won't reproduce the audio tones reliably, etc, etc. Lurking behind all of that is a defacto telephony standard that essentially supports dtmf end-to-end regardless of what type of circuit is involved (dtmf passed in-band after answer). Is it possible that part of your issue might involve the answer status of the iax path? Hi Tony I am having a similar issue to you - from the 'other' direction in that when I connect to * via IAX2 the DTMF is being ignored. I am running HEAD at the moment. (and for the benefit of another subscriber so that they don't have to invoke their autoresponder I acknowlege that DTMF is sent out of band over IAX...) http://bugs.digium.com/view.php?id=4631 Does this bear any relation? Mark On 7/12/05, Tony Mountifield [EMAIL PROTECTED] wrote: I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have the following dialplan entries, that I can dial into: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,SendDTMF(1234567890) exten = s,4,Wait(2) exten = s,5,Hangup What happens is that whatever string I give to SendDTMF, I only hear the first DTMF digit. The remaining digits don't get sent. I recently updated from an April CVS-STABLE to the July 4 version, but I couldn't see any relevant differences in the code. As its a production system, I can't just revert to test without planning. I don't know whether Voiptalk have changed anything. Is this a known bug in certain versions of Asterisk? If I do iax2 debug, I *can* see DTMF frames being sent and acked for each digit. Wo I can't understand why I'm not hearing all the digits. Any ideas? Cheers Tony ___ 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] IP Phone with Standard Power Ethernet
Kevin P. Fleming wrote: chris gamble wrote: Whats a good high quality ip phone that uses IEEE power over ethernet -- or is there a problem with IEEE power over ethernet?? Polycom IP301/IP501/IP600 all support IEEE 802.3af right out of the box. That's actually not correct. The IP600 supports PoE out of the box. The IP 30x and 50x support PoE with a special cable from Polycom. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] IPSwitchBoard shows Call Charges
Version 0.122 - 12 July 2005 * Call charges are now shown on the Calls page * IPSwitchBoard will check for a live connection every minute and reconnect if the connection is lost for some reason (asterisk restart etc.) * Bug fixes FREE Download: http://ipswitchboard.thorben.dk IPSwitchBoard is an Operators Panel for the Asterisk PBX. IPSwitchBoard is a FREE Windows.NET application which gives you: Unattended/attended transfers. Park calls and retrieve/forward them again. Organize all your SIP, IAX, CAPI and Zap extensions (automatically retrieved from Asterisk). Hotel/Call shop Billing module Monitor all extensions, queues, agents and Parked Calls. Dynamically log extensions in and out of queues. Integration with CRM software on the web. Browse Call Records and make Charts. Record calls and transfer wav files to the PC automatically. Set Do Not Disturb on Extensions and give a reason. Speed Dialing. Speed Dial Numbers can be shared from the server. ___ 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] asking again
Hi, I'm using firefly for windows (only!!) and it seems to work well...has sip and iax support, many account (useful for testing on many asterisk pbx). Giorgio. Zoa wrote: Those 2 softphones below only do SIP, no IAX. Zoa, -- www.asteriskguru.com Mohamed A. Gombolaty wrote: Hi Wasim, Check out the x-lite softphone http://www.xten.com/ As for linux check this page there are two softphones type available : http://www.iptel.org/products Thx MAG wassim Darwish wrote: ok what softphone i should use to fit windows and linux supporting iax,thanks in advance. _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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 -- Thx MAG ___ 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 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 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] Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9
Dear All, I have been running an Asterisk 0.7.1 (patched with various agent applications) server for almost 2 years. We have a data center in the USA and a call center in the UK. All calls are routed to a group of central call queues in the USA. Agents from the data center, call center and from remote locations (London, Scotland, LA, Florida, and Maine) can log in, join the call queue and pick up calls This function has worked well since implementing the system and works well using SNOM 200's (data center and call center) and SJ Phone Build 1.50.271d, Mar 11 2005. I have rebuilt an identical test environment in my test lab and I can run version 0.7.1 (patched). I log in as an agent using my softphone, make a call from a second phone, I get greeted, put in a queue, given my position, the call goes through to my soft phone, I accept the call, press # and I'm on the call. I run the upgrade to version 1.0.9 and run the same test; I get greeted, put in a queue, given my position, the call goes through to my soft phone, and I accept the call, press #... I then get a message telling me that the system saying transfer? I see nothing on the CLI except the usual waiting for '#' to acknowledge To discount the SJ Phone I installed the version of X-Ten light that some of our agents/staff use and I got the same result. I checked the DTMF setting in sio.conf and these appear correct. I downgrade to 0.7.1 and the function works on both SJ Phone and X-ten light. I have included the CLI captures below; Sip show agents; (Angela Holt) available at '[EMAIL PROTECTED]' (musiconhold is 'default') From the CLI -- outgoing agentcall, to agent '1031', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1031 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/phone6SIP/0401|20|tr) in new stack -- Called phone6 -- Called 0401 -- Agent/1031 is ringing -- SIP/phone6-1d2b is ringing -- Agent/1031 is ringing -- SIP/phone6-1d2b answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered, waiting for '#' to acknowledge I need to move to the latest version of Asterisk to enable me to measure the number of minutes a user has been held in a queue. This function was not available in version 0.7.1. I remember a similar problem with version 0.7.2. Anyone else run into the same issue? Is it a known issue/bug? What is the fix? Thanks and Regards Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ 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] monitor using incorrect path
Tzafrir Cohen wrote: Here is some logging: monitor executing ( nice -n 19 soxmix //var/spool/asterisk/monitor/SIP-242-027e_1-in.wav //var/spool/asterisk/monitor/SIP-242-027e_1-out.wav //var/spool/asterisk/monitor/SIP-242-027e_1.wav rm -f //var/spool/asterisk/monitor/SIP-242-027e_1-* ) Logging from what exactly? The output is from /var/log/asterisk/full That shouldn't be a problem on any posix system (except cygwin) . '//' is simply translated to '/' . I suspect you have a different problem. Indeed, when I do this manually (by using soxmix on command line) this works. Any idea why this doesn't work when this gets executed after the hangup? Cheers, Kristof. ___ 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] How to avoid collect calls on ISDN BRI trunks?
Hello, friends. I need to block collect calls on my PBX. I was able to find information on Google regarding ISDN ZAP channels, but not ISDN CAPI channels which is my case. Since there's no information from the Telco that the call is going to be charged by the callee, if a particular call is automatically answered by the PBX instead of a real person there's no way to avoid it. There's only a recording AFTER the call is answered, asking for the callee party to accept it. By using ZAP channels with HFC ISDN cards, it's possible to use Hangup() and Flash() commands, which won't do any good on CAPI channels because it will hangup the call immediatelly. Note: This situation is exctaly the same with remote access servers (RAS), commonly used by Internet Service Providers. Here's an example of what should happen on my scenario: [default] ; External calls comes on default context exten = s,1,Wait,1 ; Wait for all ISDN and CAPI messages exten = s,2,Answer() ; PBX actually answers the call exten = s,3,Wait,1 ; Just in case exten = s,4,Flash() ; Avoid collect calls, don't actually hangup exten = s,5,Wait,1 ; Wait a second (just in case) exten = s,6,Answer() ; Now it should work as if it were s,1 exten = s,7,BackGround(IVR-menu) ; 1 for sales, 2 for support, 3 for... Note: Step s,4 could be also Hangup(), it won't make any difference. If it were an analog trunk, it would work flawlessly. Unfortunatelly for digital trunks, when you hangup the channel the call is dropped immediately. I'm pretty sure it's not an Asterisk issue, but it's definitely related. Thanks in advance for any input on this matter. Regards, -Dhennys ___ 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] Pushing new firmware to Snom 190
Take a look at the advanced page. At the bottom is an option. Set it to update automatically, then the phone will not wait for user interaction on boot-up when a new firmware is available. Regards Nils Ohlmeier On Monday 11 July 2005 23:27, Colin Anderson wrote: Anyone know how I can push a firmware update to a Snom 190 without using DHCP? In the web interface, I specify a path to the Snom firmware, and it works, except I have to physically press OK to get the update to download. I need to do it remotely... ___ 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 -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.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] Re: DTMF not sending properly via IAX
In article [EMAIL PROTECTED], Rich Adamson [EMAIL PROTECTED] wrote: Not sure this applies, but a few months ago some of us were having problems with an itsp (livevoip.com) and incoming iax calls that hit the * ivr. dtmf was essentially not being passed. I opened a bug, however Mark quickly closed it with a note that's the way iax works; have the itsp fix their problem. The issue at that time was oriented around a call that was considered answered. If the end-to-end path of the call is considered answered, asterisk is no longer in the voice path to translate dtmf into iax out-of-band dtmf packets. Part of the argument was then what is an asterisk user suppose to do to pass dtmf across an iax link that uses g729? g729 won't reproduce the audio tones reliably, etc, etc. Lurking behind all of that is a defacto telephony standard that essentially supports dtmf end-to-end regardless of what type of circuit is involved (dtmf passed in-band after answer). Is it possible that part of your issue might involve the answer status of the iax path? I doubt it. The puzzling thing is that it used to work, and doesn't now. So it is either something I've changed (I updated from April CVS-STABLE to July CVS-STABLE), or something the provider has changed (I have raised a ticket with them). I'm not sure about your second paragraph. As I understand it, a device that interfaces from TDM to IAX is supposed to detect DTMF tones, and translate them into out-of-band IAX DTMF control frames. Conversely a device going from IAX to TDM should generate outgoing inband tones when it gets such DTMF control frames. Surely if Asterisk is interfacing between IAX and TDM, it must remain in the voice path all the time? The salient feature of my current problem is that the first digit of a SendDTMF string does get sent out inband by my provider, but subsequent digits don't. My feeling is that it has something to do with seqnos or timestamps in the DTMF control frames, but I don't know what. I've even tried a loop in the dialplan to peel off the digits one by one and send them individually using SendDTMF with a Wait(1) in between, but it doesn't seem to help. However, after some other audio activity and a delay, a fresh SendDTMF will again have its first digit converted to tones, but not subsequent digits. Bizarre! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] How to avoid collect calls on ISDN BRI trunks?
On Tue, 12 Jul 2005, Dhennys Pestana wrote: I need to block collect calls on my PBX. I was able to find information on Google regarding ISDN ZAP channels, but not ISDN CAPI channels which is my case. Since there's no information from the Telco that the call is going to be charged by the callee, if a particular call is automatically answered by the PBX instead of a real person there's no way to avoid it. There's only a recording AFTER the call is answered, asking for the callee party to accept it. By using ZAP channels with HFC ISDN cards, it's possible to use Hangup() and Flash() commands, which won't do any good on CAPI channels because it will hangup the call immediatelly. Note: This situation is exctaly the same with remote access servers (RAS), commonly used by Internet Service Providers. Here's an example of what should happen on my scenario: [default] ; External calls comes on default context exten = s,1,Wait,1 ; Wait for all ISDN and CAPI messages exten = s,2,Answer() ; PBX actually answers the call exten = s,3,Wait,1 ; Just in case exten = s,4,Flash() ; Avoid collect calls, don't actually hangup exten = s,5,Wait,1 ; Wait a second (just in case) exten = s,6,Answer() ; Now it should work as if it were s,1 exten = s,7,BackGround(IVR-menu) ; 1 for sales, 2 for support, 3 for... Note: Step s,4 could be also Hangup(), it won't make any difference. If it were an analog trunk, it would work flawlessly. Unfortunatelly for digital trunks, when you hangup the channel the call is dropped immediately. Yes, because Hangup() means 'disconnect'. I'm not aware of Flash() and what it is doing, but it is surely not implemented in chan_capi. If someone can tell me what Flash() is supposed to do, we can implement it in chan_capi-cm. Armin ___ 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] Help: TE100P connecting to non PRI, ISDN interfaces
Hello, i've googled and can't find a definite answer, so here goes: I have purchased the Digium TE100P, and am setting up the connection, however the telco i'm supposed to work with does not support PRI/ISDN E1 connections. They only support E1/R2 lines. Is there a way i can make the TE100P work with this? I've not seen any zaptel.conf that supports this. Any workarounds? Thanks for any help! Ian ___ 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] How to avoid collect calls on ISDN BRI trunks?
Dhennys, I would expect that the ISDN collect call would have some kind of notification about the charge. In E1/R2, the Telebras standard in fact DOES have this notification defined, from what I remember, the problem was that many of the CO switches would not support it, that is why the 'double-answering' is used in most if not all the cases (there is a condition where the double-answering would tear down any call, something to do with the configuration in the PSTN switch). I don't think you can answer/hangup/answer a call in ISDN the way you can in E1/R2 or analog lines... Armin Schindler wrote: On Tue, 12 Jul 2005, Dhennys Pestana wrote: I need to block collect calls on my PBX. I was able to find information on Google regarding ISDN ZAP channels, but not ISDN CAPI channels which is my case. Since there's no information from the Telco that the call is going to be charged by the callee, if a particular call is automatically answered by the PBX instead of a real person there's no way to avoid it. There's only a recording AFTER the call is answered, asking for the callee party to accept it. By using ZAP channels with HFC ISDN cards, it's possible to use Hangup() and Flash() commands, which won't do any good on CAPI channels because it will hangup the call immediatelly. Note: This situation is exctaly the same with remote access servers (RAS), commonly used by Internet Service Providers. Here's an example of what should happen on my scenario: [default] ; External calls comes on default context exten = s,1,Wait,1 ; Wait for all ISDN and CAPI messages exten = s,2,Answer() ; PBX actually answers the call exten = s,3,Wait,1 ; Just in case exten = s,4,Flash() ; Avoid collect calls, don't actually hangup exten = s,5,Wait,1 ; Wait a second (just in case) exten = s,6,Answer() ; Now it should work as if it were s,1 exten = s,7,BackGround(IVR-menu) ; 1 for sales, 2 for support, 3 for... Note: Step s,4 could be also Hangup(), it won't make any difference. If it were an analog trunk, it would work flawlessly. Unfortunatelly for digital trunks, when you hangup the channel the call is dropped immediately. Yes, because Hangup() means 'disconnect'. I'm not aware of Flash() and what it is doing, but it is surely not implemented in chan_capi. If someone can tell me what Flash() is supposed to do, we can implement it in chan_capi-cm. ___ 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] Will pay for asterisk help...
Hello Guys... I'm looking for someone to help me out with an [EMAIL PROTECTED] installation... I've managed to get it working to the point that extensions can talk to each other, but not incoming calls or outgoing calls... I need to get it configured with Broadvoice... I'm willing to pay someone to help me do this... Contact me at [EMAIL PROTECTED] if you're interested. Thanks ___ 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] Snom 360 NOTIFY syntax
On Mon, Jul 11, 2005 at 01:16:08PM -0500, Patrick Friedel wrote: I'm rolling out an installation with snom 360s in the near future. Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002. I have the 360's set up to subscribe and notify for the line use lights, which works like a charm for interoffice calling (between the 360's, anyway. The IAXy, 200 and, softphone will be used by less phone dependant types) but what I can't figure out from the Wiki is if it's possible to have the ZAP lines notify for the outbound lines so we can see how many lines are in use. I am by no means an expert at this, but I did some experimentation and it appears that the NOTIFY will not get sent for the trunk lines, only for extensions. I also found that the SUBSCRIBE/NOTIFY sequence only works for SIP and ZAP, I couldn't get it to work for IAX2. I do not know why this would be, and it is possible I was doing something wrong, but for what it's worth, that's my experience so far. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Will pay for asterisk help...
It would be better if you could give us some more details about your configuration so that someone could help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 3:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Will pay for asterisk help... Hello Guys... I'm looking for someone to help me out with an [EMAIL PROTECTED] installation... I've managed to get it working to the point that extensions can talk to each other, but not incoming calls or outgoing calls... I need to get it configured with Broadvoice... I'm willing to pay someone to help me do this... Contact me at [EMAIL PROTECTED] if you're interested. Thanks ___ 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 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] choosing a softphone
which the best softphone that works with window and linux supporting IAX2 ,thanks in advance. __ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ 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] choosing a softphone
which is the best softphone that works with window and linux supporting IAX2 ,thanks in advance. Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.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
Re: [Asterisk-Users] choosing a softphone
jonny hashem wrote: which the best softphone that works with window and linux supporting IAX2 ,thanks in advance. None. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] how to debug perl agi
hello
i am trying to develop perl application for asterisk
with radius accounting how can i debug that weather
callback is working when call is stoped.
how can i check this
syslog('info', 'hello Asterisk!');
thanks
Kamran
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http://auctions.yahoo.com/
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Re: [Asterisk-Users] Unable to dial certain calls
Of course. Note that I have no idea what glaw is but someone on some board shttps://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gif https://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gifuggested it as a resolution to a similiar problem so I put it in. The entry from the iax.conf file is: [vbx] type=peer host= 213.61.187.150 secret=-my password- notransfer=yes context=def allow=glaw What is glaw? allow=ulaw allow=gsm I'd suggest changing the above to something like this: disallow=all allow=ulaw allow=gsm and try again. Also, check the syntax throughout your files as it appears that same undefined glaw appears elsewhere. ___ 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] Enabling rtcachefriends prevents phones from calling each other
Kevin P. Fleming wrote: Matthew Boehm wrote: Can't it be changed so that if Server A has stored an unknown address for phone B that if it needs to contact B again it should look up in the database to try and contact it instead of just giving up? Perhaps rtagressive option? Contact only, not storing info in cache. Can it be done? Of course, it's all just code :-) I think it would be reasonable to add that as an option, but the number of Realtime-related options is rapidly getting out of hand and people will not be able to understand what they all do and how they interact... Please, please add this option. If you send me a patch, I will gladly volunteer to test it thoroughly. Having both MWI working and multiple servers working is a must for us. Thanks much, Rana Dutt ___ 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: DTMF not sending properly via IAX
I'd be interested to understand where you read about the 'sequence numbers' issue. This sounds like it might relate to the problem I am experiencing. It's either that, or the DTMF is coming inband over the IAX channel... ;-) Mark On 7/12/05, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED] ,Mark Edwards [EMAIL PROTECTED] wrote: Hi TonyI am having a similar issue to you - from the 'other' direction in that when I connect to * via IAX2 the DTMF is being ignored. I am running HEAD at the moment.(and for the benefit of another subscriber so that they don't have to invoke their autoresponder I acknowlege that DTMF is sent out of band over IAX...) http://bugs.digium.com/view.php?id=4631Does this bear any relation?I'm not sure. Incoming DTMF using IAX2 from the PSTN via my provider to mybox seems to work fine. When my box tries to send DTMF, it appears that the first digit of the string is translated by the provider into DTMF tonesover the PSTN, but then subsequent digits in the same string, although sentby IAX2 from my box, are not translated by the provider. I remember reading something some time ago about DTMF getting lost due tosequence number problems, but I can't remember whether that was SIP or IAX.I'm hoping to avoid having to delve into the code myself, due to lack of time, and I'm not even sure whether the problem is my end or the provider's.CheersTonyMarkOn 7/12/05, Tony Mountifield [EMAIL PROTECTED] wrote: I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have the following dialplan entries, that I can dial into: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,SendDTMF(1234567890) exten = s,4,Wait(2) exten = s,5,Hangup What happens is that whatever string I give to SendDTMF, I only hear the first DTMF digit. The remaining digits don't get sent. I recently updated from an April CVS-STABLE to the July 4 version, but I couldn't see any relevant differences in the code. As its a production system, I can't just revert to test without planning. I don't know whether Voiptalk have changed anything. Is this a known bug in certain versions of Asterisk? If I do iax2 debug, I *can* see DTMF frames being sent and acked for each digit. Wo I can't understand why I'm not hearing all the digits.--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-- regards, Mark P. EdwardsFWD: 667917 ___ 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] IP Phone with Standard Power Ethernet
Hi, OptiPoint 4x0/600 supports IEEE 802.3af. --- chris gamble [EMAIL PROTECTED] wrote: I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip phone that uses IEEE power over ethernet -- or is there a problem with IEEE power over ethernet?? ___ 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 __ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ 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] Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9
I get greeted, put in a queue, given my position, the call goes through to my soft phone, and I accept the call, press #... I then get a message telling me that the system saying transfer? I see nothing on the CLI except the usual waiting for '#' to acknowledge Send the complete extensions.conf for the incoming call portion, and the agentcallbacklogin section. Also send the complete CLI from the call arriving into the PABX through to the call being sent to the agent. I suspect somewhere you are including the t or T option to the queue or dial which allows # to transfer a call. Of course, perhaps someone should check this, as we can't transfer a call until after we accept it... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Referrals/Success Stories would be greatly appreciated
Hello, I am looking to replace my company's Avaya Merlin Magix system with an Asterisk based PBX when our current lease is up. I had a meeting with upper management yesterday, and they would like some assurance that other companies are running Asterisk with success. We are a relatively small company, with about 70 total extensions. I would be purchasing a Dell Poweredge 2850 Server and most likely a Sangoma A101U. The phone service would be provided by a ISDN PRI from SBC, with Caller ID and DID. The phones we are looking at are the Uniden UIP-200, and the Cisco 7940G on the high end. I've become fairly familiar with Asterisk, and i'm pretty confident that I should be able to squash most of the problems that come up with the help of Google. If anyone would be so kind as to post success stories, or even potential problems we should look out for, I would greatly appreciate it. Thank you, Jock W. Shirey Network Administrator Dowding Industries, Inc. ___ 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] Snom phones - any advice
We initially tried going with Snom but had a high failure rate. Since then we have been Polycom and they seem to work great. We did just have a customer demand a phone with a sidecar and bought a Snom 220 with 2 of them. The additional sidecars require a seperate power supply which did not come with. Also they claim damage if not powered up properly. After installed for a week it too had issues. Seems the handset went bad which also made the speakerphone stop working. I would strongly look at Polycom or Cisco if this is a business class rollout. We did also test these but they were too expensive. Good Luck On Jul 6, 2005, at 11:31 AM, Patrick Fortin wrote: Hi We are about to buy several Snom phones. Does anyone have warnings or advices against these phones ? Our finalists were Cisco, Polycom and Snom. We will be using only the SIP protocol. Thanks Patrick ___ 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 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] How to integrate the Call Pickup with CID info feature in the release tree of Asterisk?
Hi, we really need the feature Call Pickup with CID info http://www.voip-info.org/wiki-Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP in the current Asterisk release because we have a newer TE405P card which needs 1.0.8 or newer to work. The call pickup patch only works for 1.0.7. Who is responsible for such a wish? Regards, Kib ___ 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] Enabling rtcachefriends prevents phones from calling each other
Rana Dutt wrote: Please, please add this option. If you send me a patch, I will gladly volunteer to test it thoroughly. I did not volunteer to write it, only commented that it would be an acceptable option to add. My project list is already quite long :-) Having both MWI working and multiple servers working is a must for us. There are developers who write Asterisk code who would gladly do this work for you, for a fee, if it's a 'must' for you. Otherwise you'll just have to wait until someone finds it important enough for their own systems to do the work. ___ 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] IP Phone with Standard Power Ethernet
Eric Wieling aka ManxPower wrote: That's actually not correct. The IP600 supports PoE out of the box. The IP 30x and 50x support PoE with a special cable from Polycom. Bummer...I thought the built-in PoE chip was one of the few upgrades in the 300-301 and 500-501 paths... too bad, would have been a good thing to include. ___ 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] Re: DTMF not sending properly via IAX
In article [EMAIL PROTECTED], Mark Edwards [EMAIL PROTECTED] wrote: I'd be interested to understand where you read about the 'sequence numbers' issue. This sounds like it might relate to the problem I am experiencing. It's either that, or the DTMF is coming inband over the IAX channel... ;-) Well you could try doing a tcpdump to capture the incoming and outgoing packets on the IAX port. Use the -w option with -s0, to write the whole of each packet to the packet file. You could then use a recent version of Ethereal to read the packet file and analyse it. Apparently, recent versions of Ethereal understand IAX. Ethereal is able to save RTP audio streams as audio files - it might be able to do the same with IAX audio streams, but I don't know. It was probably on the asterisk-dev list where I read about sequence number issues, but I can't remember how long ago. I'm not even sure whether it was IAX or SIP. I've just looked through the cvs log of chan_iax2.c and chan_sip.c, but didn't find anything that appeared relevant. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Last CVS - High Load
Yes I have experienced the same on my test machine. It has been like this for 3 weeks of CVS Head. Someone must havea look at that, I think is the SIP channel. - Original Message - From: Thierry Wehr To: asterisk-users@lists.digium.com Sent: Tuesday, July 12, 2005 10:37 AM Subject: [Asterisk-Users] Last CVS - High Load Good morning on our Test Machine based on RedHat 9 on a Pentium 4 SMP we are experiencing a 99% CPU load for asterisk with the CVS of tonight does anyone noticed that best regards Thierry ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] IP Phone with Standard Power Ethernet
The statements about Cisco and PoE aren't strictly correct. Cisco 7940 and 7960 phones without the G (global) suffix used Cisco PoE and had their keys labled in local languages. The 7940G and 7960G Global phones are IEEE 802.3af PoE and have the keys engraved with icons and stick over labels. Cisco PoE switches like the C3560-48-PS support both Cisco PoE and IEEE 802.3af PoE automagically. The Cisco 7912G supports IEEE 802.3af PoE and is a very nice singel line phone - I have about 8 dotted around the internet at various friends and family. Mike Hi, OptiPoint 4x0/600 supports IEEE 802.3af. --- chris gamble [EMAIL PROTECTED] wrote: I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip phone that uses IEEE power over ethernet -- or is there a problem with IEEE power over ethernet?? ___ 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] NAT=YES
Title: NAT=YES Good morning Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP. Configuration Info: I have all users in SIP.CONF configured with NAT=YES Asterisk has a public IP Remote host is behind a firewall with NAT When I sniff on the Asterisk public network, I see the following. 1. INVITE from remote host public IP to Asterisk public IP 2. 183 response from Asterisk public IP to remote host public IP 3. RTP from Asterisk public IP to the remote host private IP 4. RTP from remote host public IP to Asterisk public IP 5. RTP from Asterisk public IP to the remote host public IP Is there a way to prevent step 3 from happening? Or, is there a way to delay the invalid RTP from being sent from the Asterisk in step 3? Does anyone know why the Asterisk sends RTP to remote host private IP? I would expect NAT=YES to correct this issue. Thanks, Peter ___ 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] New Cisco 7960 Firmware 7.5
Hi, new features: RFC 3261 compliance (no TCP) RFC 3264 compliance RFC 3311 Compliance (display updates only, no media) Remote-Party-ID for display updatesA Remote-Party-ID header received in an INVITE or 200 OK will now update the display of the phone to accurately reflect the connected party New Configuration parameters sip_max_forwards and rfc_2543_hold REGISTER contact header sip.instance parameter support Does asterisk allready support supervised-transfers-with-correct-number (c sees number of a after b, who transferred a to c, hung up)...? Any other ideas what could be done with RFC3311/Remote-Party-ID-updates? Regards aa _ Need a new job? Check out XtraMSN Careers http://xtramsn.co.nz/careers ___ 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] Will pay for asterisk help...
I can help you! I can be reach at this email [EMAIL PROTECTED] or via IM Yahoo jraborg, no problem, what kind of FXS or FXO are you using? JR It would be better if you could give us some more details about your configuration so that someone could help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 3:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Will pay for asterisk help... Hello Guys... I'm looking for someone to help me out with an [EMAIL PROTECTED] installation... I've managed to get it working to the point that extensions can talk to each other, but not incoming calls or outgoing calls... I need to get it configured with Broadvoice... I'm willing to pay someone to help me do this... Contact me at [EMAIL PROTECTED] if you're interested. Thanks ___ 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 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 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 realtime failover problems
Mohamed A. Gombolaty wrote: Dear All, I was trying to use Realtime Asterisk option for Sip users and peers + Heartbeat + Mysql Replication in order to make a failover system, so that if Ast1 went down for any reason, Ast2 server will have the same data that Ast1 used in the Mysql database and don't need to make the phones re-register. But when I started testing: the calls that where active during the transition between the two servers where disconnected (the two phones are talking peer to peer thanks to the canreinvite option but they we still sending UDP packets to port 1025 to the asterisk server), the phones must re-register with the new server though the Mysql server was replicated and the new server should have the data it needs. Has anyone trid doing this before, or does anyone have any idea if this should work or is there another way to do so, I will really appreciate it very much if anyone has any helping pointers. It was not necessary for you to post twice. The behavior you are looking for is also what someone else is looking for. Look for this subject Enabling rtcachefriends prevents phones from calling each other in your emails. Perhaps you and Rana can put together a bounty for this project. I'll gladly accept it. -Matthew ___ 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] Help: TE100P connecting to non PRI, ISDN interfaces
What king of signaling your telco support? try on /etc/zaptel.conf span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 I have it working with : switchtype = euroisdn signalling = pri_cpe Cheers. JR Hello, i've googled and can't find a definite answer, so here goes: I have purchased the Digium TE100P, and am setting up the connection, however the telco i'm supposed to work with does not support PRI/ISDN E1 connections. They only support E1/R2 lines. Is there a way i can make the TE100P work with this? I've not seen any zaptel.conf that supports this. Any workarounds? Thanks for any help! Ian ___ 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 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] IP Phone with Standard Power Ethernet
Michael J. Tubby B.Sc (Hons) G8TIC wrote: Cisco 7940 and 7960 phones without the G (global) suffix used Cisco PoE and had their keys labled in local languages. The 7940G and 7960G Global phones are IEEE 802.3af PoE and have the keys engraved with icons and stick over labels. Are you sure about that? I have users with 7960G phones (icon buttons) that did not work without making cross-over patch cables from our PoE injectors. ___ 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] bristuff patches and realtime mysql
Christoph wrote: Hi! I have a problem compiling the res_config_mysql.so after successfully compiling and installing asterisk with the bristuff package/patches. I get lots of compiler errors. When inserting a previously compiled res_config_mysql.so into the bristuff-patched Asterisk, I get an error on startup and Asterisk crashes. Anyone using res_config_mysql.so with the bristuff package and can help me? Thanks, Christoph How do you expect me to possibly fix my module if you don't supply any compile errors? I don't use BRI so you will need to provide me alot of info. Which bri package did you install? Perhaps I can install that in a tmp dir and see what results I get. -Matthew ___ 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] Enabling rtcachefriends prevents phones from calling each other
Kevin P. Fleming wrote: Rana Dutt wrote: Please, please add this option. If you send me a patch, I will gladly volunteer to test it thoroughly. I did not volunteer to write it, only commented that it would be an acceptable option to add. My project list is already quite long :-) Having both MWI working and multiple servers working is a must for us. There are developers who write Asterisk code who would gladly do this work for you, for a fee, if it's a 'must' for you. Otherwise you'll just have to wait until someone finds it important enough for their own systems to do the work. There is another person on the list looking for this as well, though he is asking slightly differently. Perhaps you two could put up a bounty. I'll gladly accept. ;) -Matthew ___ 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] Unable to dial certain calls
First off kill the Glaw. It doesn't exist.
Then try your call. But also why are you sending the line congestion
when you first start to make a call. That's normally used as a closure.
But from what I can see about the only thing wrong is the GLAW. Kill
that and you should be good to go.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JP Russell
Sent: Tuesday, July 12, 2005 5:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unable to dial certain calls
Of course. Note that I have no idea what glaw is but
someone on some board
shttps://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gif
https://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gifugge
sted
it as a resolution to a similiar problem so I put it in.
The entry from the iax.conf file is:
[vbx]
type=peer
host= 213.61.187.150
secret=-my password-
notransfer=yes
context=def
allow=glaw
allow=ulaw
allow=gsm
and from extensions.conf I guess you need the [def]
context entries.
they are:
;NL
exten = _00316.,1,Congestion
exten = _00319.,1,Congestion
exten = _0031X.,1,SetCallerID(Not Available
7005551212)
exten = _0031X.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _0031X.,3,Playback(invalid)
exten = _0031X.,4,Hangup
;US
exten = _001X.,1,SetCallerID(Not Available
7005551212)
exten = _001X.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _001X.,3,Playback(invalid)
exten = _001X.,4,Hangup
Finally sip.conf includes the below paramaters:
[general]
disallow=all
allow=ulaw
allow=glaw
allow=gsm
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = from-sip ; Default for incoming calls
callerid=No CallID
[2203]
port=5061
username=-thisusername-
secret=-this password-
host=dynamic
type=friend
nat=1
qualify=no
;reinvite=no
canreinvite=yes
context=intern
On Mon, 11 Jul 2005 22:55:49 -0400
Brian C. Fertig [EMAIL PROTECTED] wrote:
Check your codecs.. Can you post a sniplet of your IAX,
SIP, and extensions.conf for dialing the US so we can see
were the problem may lie?
Brian Fertig
From: [EMAIL PROTECTED] on behalf
of JP Russell
Sent: Mon 7/11/2005 9:12 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to dial certain calls
To begin with, I am new to both asterisk and VOIP and
although I've
gotten pretty far with my Asterisk setup and have two
different sip
accounts working fine for outgoing calls I am having
trouble with one
issue.
My problem is that I have another provider who uses IAX2
that I wish
to use for calling various countries, including local
(The
Netherlands) calls and calls to the US to name two. I
am able to
call local numbers without a problem through this
provider with
Asterisk, but calling US numbers is not working.
I CAN call the same US numbers with the service by using
a direct
connection from a softphone for example.
The entries that show up in the log after failed
attempts to call the
US are:
Jul 11 20:04:04 WARNING[25225728]: File channel.c, Line
1851
(ast_channel_make_compatible): No path to translate from
SIP/2203-2929
(4) to IAX2[vbx]/1(16)
Jul 11 20:04:04 WARNING[25225728]: File app_dial.c, Line
672
(dial_exec): Had to drop call because I couldn't make
SIP/2203-2929
compatible with IAX2[vbx]/1
I don't see anything suspicious entries in the CLI
logging with IAX2
debugging on. Searching the archives and google didn't
turn up a
solution to this or even point me in the right direction
I'm afraid.
Anyone have any idea on what my problem is or I can go
for this issue?
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RE: [Asterisk-Users] Pushing new firmware to Snom 190
Thanks for replying. Frustrating, didn't work. Set it to update automatically, and made an HTML page consisting of: html pre bootloader: firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin /pre /html Added the HTML page to a webserver I control, and added it's URL to the Setting URL field. Rebooted. Didn't update. Also added http://www.snom.com/download/snom190-3.56m-SIP-j.bin to Software Update Firmware field, hit Load, nothing. System Information yields: Version-Code: snom190-SIP 3.44 -but- Firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin Phones that I have sucessfully updated when I am physically present to press the OK button yield: Version-Code: snom190-SIP 3.56m Firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin Anyone got any other ideas? tia -Original Message- From: Nils Ohlmeier [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 5:40 AM To: asterisk-users@lists.digium.com Cc: Colin Anderson Subject: Re: [Asterisk-Users] Pushing new firmware to Snom 190 Take a look at the advanced page. At the bottom is an option. Set it to update automatically, then the phone will not wait for user interaction on boot-up when a new firmware is available. Regards Nils Ohlmeier On Monday 11 July 2005 23:27, Colin Anderson wrote: Anyone know how I can push a firmware update to a Snom 190 without using DHCP? In the web interface, I specify a path to the Snom firmware, and it works, except I have to physically press OK to get the update to download. I need to do it remotely... ___ 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 -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.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] PRI problem
currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet.here si the sip debug msg, we got a Message type: DISCONNECT (69) and unable to dial any numbers.Jul 12 12:56:26 WARNING[1523]: chan_zap.c:1931 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16!-- Making new call for cr 32771 Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 04 e9 81 83 81] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0ChanSel: Reserved Ext: 1 DS1 Identifier: 1 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 08 4a 69 61 6e 20 4c 69 75] Display (len= 8) [ Jian Liu ] [6c 04 21 81 31 30] Calling Number (len= 6) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '10' ] [70 0d a1 30 31 33 39 30 31 30 33 35 34 33 36] Called Number (len=15) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '013901035436' ]NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request, peerstate Disconnect Indication 需要一个2000兆的免费邮箱吗？网易免费邮箱是中国最多人使用的电子邮箱。 ___ 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] bristuff patches and realtime mysql
On Tue, 2005-07-12 at 09:42 -0500, Matthew Boehm wrote: How do you expect me to possibly fix my module if you don't supply any compile errors? I don't use BRI so you will need to provide me alot of info. Which bri package did you install? Perhaps I can install that in a tmp dir and see what results I get. -Matthew Hi Matthew, I didn't want to spam the mailinglist with all those error messages generated by the compiler. I used the latest bri package from junghanns.net: http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC8h.tar.gz The package then downloads libpri-1.0.8, zaptel-1.0.8, cwain and asterisk-1.0.8 and does a lot of patching to the sources. This is then the error message that I get when trying to compile the asterisk-addons: pound:/usr/src/asterisk-addons# make all cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_saycountpl.o app_saycountpl.c cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o 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: In function `my_load_module': cdr_addon_mysql.c:268: warning: assignment makes pointer from integer without a cast cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 It would be really great if you could get it to work as I think it's an excellent addition for Asterisk. If you need more info just write me a message, Christoph ___ 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] IP Phone with Standard Power Ethernet
We sell an 802.3AF PoE Injector that includes a standard RJ45 patch cable and the reverse polarity RJ45 patch cables for use with Cisco PoE Endpoints. Cost is $29.95/ea quantity discounts are available. http://www.voipsupply.com/product_info.php?manufacturers_id=22products_id=570 Cory Andrews Purchasing / EVP VOIPSupply.com v – 716.630.1555 X22 e – [EMAIL PROTECTED] Tom wrote: At 09:31 PM 7/11/2005, you wrote: I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip phone that uses IEEE power over ethernet -- or is there a problem with IEEE power over ethernet?? Cisco supports 802.3af with a special cross-over cable which is easy and cheap to make. http://www.voip-info.org/tiki-index.php?page=Cisco+POE We are using this with both the 7905g and 7960g phones. We are quite happy with the phones. Tom ___ 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 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] Cisco 7960 firmware upgrade promblems
Sergio Chersovani wrote: I know it's hard to find out infos at the cisco site. Maybe you can open a TAC case Sergio I did find this info: http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20cisco%2079xx comments_threshold=0comments_offset=0comments_sort_mode=commentDate_desc comments_maxComments=10comments_parentId=353#threadId358 snip The two phones I purchased had Application Load ID (AKA: firmware) of P003AM30. This is their skinny protocol load. If you're trying to do sIP, you need a load that starts out POS.. You can not upgrade from P00 to P0S, you need to downgrade to P0S30203 to get it using POS firmware, then you can upgrade to the newer releases of the SIP firmware, with one extra thing to know. You do not need to step through every version of he firmware, you can jump versions of firmware, but what you encouter is the issue with their signed binaries (ie: *.sbn files) that they have converted to. If you have both a *.bin and a *.sbn file in the TFTP server root directory, it will default to loading the *.bin (ie: unsigned binary), which you do not want to do, since you need to convert over to signed binaries, in order to continue upgrading to get to the higher versions which only come signed. If you try to load higher version binaries that are not signed, the phone will fail to load and give an error as such (which I dont have the exact verbiage of). So, bottomline, go down to SIP 2.3, then go up to the first signed binary, then go to the final signed binary, then you ought to be there. /snip ___ 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] NAT=YES
Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf Mark Klint, Peter wrote: Good morning Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP. Configuration Info: I have all users in SIP.CONF configured with NAT=YES Asterisk has a public IP Remote host is behind a firewall with NAT When I sniff on the Asterisk public network, I see the following. 1. INVITE from remote host public IP to Asterisk public IP 2. 183 response from Asterisk public IP to remote host public IP 3. RTP from Asterisk public IP to the remote host private IP 4. RTP from remote host public IP to Asterisk public IP 5. RTP from Asterisk public IP to the remote host public IP Is there a way to prevent step 3 from happening? Or, is there a way to delay the invalid RTP from being sent from the Asterisk in step 3? Does anyone know why the Asterisk sends RTP to remote host private IP? I would expect NAT=YES to correct this issue. Thanks, Peter ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
Re: [Asterisk-Users] Unable to dial certain calls
Thanks for the response. I added the glaw when it wasn't working so just removing it won't resolve the issue. I have the congestion entries in there to prevent dialing to certain types of paid lines and mobile phones in The Netherlands. I will route mobile phone calls to another provider at a later stage. Nonetheless, calls to The Netherlands are working properly, it is calls to the US that are not going through. I will remove the glaw line as you suggest and also add disallow=all and allow=gsm as Rich Adamson suggested, perhaps all together it will help. On Tue, 12 Jul 2005 10:49:11 -0400 Brian C. Fertig [EMAIL PROTECTED] wrote: First off kill the Glaw. It doesn't exist. Then try your call. But also why are you sending the line congestion when you first start to make a call. That's normally used as a closure. But from what I can see about the only thing wrong is the GLAW. Kill that and you should be good to go. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office ___ 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 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] Pushing new firmware to Snom 190
The problem with updating the firmware via settings is, that the information about the firmware are not allow in the first settings file. Instead the settings file which you entered in Setting URL needs to have the following link: firmware_status: http://your-server/which/contains/the/page/below.html The link should point to the file which copied below. If you restart the phone afterwards it should do a reboot and upgrade the firmware later automatically. Regards Nils On Tuesday 12 July 2005 16:51, Colin Anderson wrote: Thanks for replying. Frustrating, didn't work. Set it to update automatically, and made an HTML page consisting of: html pre bootloader: firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin /pre /html Added the HTML page to a webserver I control, and added it's URL to the Setting URL field. Rebooted. Didn't update. Also added http://www.snom.com/download/snom190-3.56m-SIP-j.bin to Software Update Firmware field, hit Load, nothing. System Information yields: Version-Code: snom190-SIP 3.44 -but- Firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin Phones that I have sucessfully updated when I am physically present to press the OK button yield: Version-Code: snom190-SIP 3.56m Firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin Anyone got any other ideas? tia -Original Message- From: Nils Ohlmeier [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 12, 2005 5:40 AM To: asterisk-users@lists.digium.com Cc: Colin Anderson Subject: Re: [Asterisk-Users] Pushing new firmware to Snom 190 Take a look at the advanced page. At the bottom is an option. Set it to update automatically, then the phone will not wait for user interaction on boot-up when a new firmware is available. Regards Nils Ohlmeier On Monday 11 July 2005 23:27, Colin Anderson wrote: Anyone know how I can push a firmware update to a Snom 190 without using DHCP? In the web interface, I specify a path to the Snom firmware, and it works, except I have to physically press OK to get the update to download. I need to do it remotely... ___ 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 -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.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] TDM22B - asterisk and seimens hipath 3750
Hello I am planning to build a small PBX using TDM22B. We have a Siemens Hipath 3750 in operation already. When I manage to complete my PBX using TDM22B I would ofcourse like to be able to connect my Asterisks PBX with the Siemens Hipath 3750 PBX. Will there be any issues regarding my plan ? Or is there any other issues that I need to take into account vis-a-vis Siemens PBX. I have never done all this before so I would appreciate all inputs. Thanks in advance Varun ___ 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] IP Phone with Standard Power Ethernet
Uniden, Polycom, to name 2 more. On 7/12/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote: The statements about Cisco and PoE aren't strictly correct. Cisco 7940 and 7960 phones without the G (global) suffix used Cisco PoE and had their keys labled in local languages. The 7940G and 7960G Global phones are IEEE 802.3af PoE and have the keys engraved with icons and stick over labels. Cisco PoE switches like the C3560-48-PS support both Cisco PoE and IEEE 802.3af PoE automagically. The Cisco 7912G supports IEEE 802.3af PoE and is a very nice singel line phone - I have about 8 dotted around the internet at various friends and family. Mike Hi, OptiPoint 4x0/600 supports IEEE 802.3af. --- chris gamble [EMAIL PROTECTED] wrote: I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip phone that uses IEEE power over ethernet -- or is there a problem with IEEE power over ethernet?? ___ 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 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] NAT=YES
FYI, there is no such thing as reinvite. Someone started using that in postings a long time ago and a few people keep posting it, but it doesn't exist. (Check /usr/src/astersik/configs/sip.conf.sample) Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf Mark Klint, Peter wrote: Good morning Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP. Configuration Info: I have all users in SIP.CONF configured with NAT=YES Asterisk has a public IP Remote host is behind a firewall with NAT When I sniff on the Asterisk public network, I see the following. 1. INVITE from remote host public IP to Asterisk public IP 2. 183 response from Asterisk public IP to remote host public IP 3. RTP from Asterisk public IP to the remote host private IP 4. RTP from remote host public IP to Asterisk public IP 5. RTP from Asterisk public IP to the remote host public IP Is there a way to prevent step 3 from happening? Or, is there a way to delay the invalid RTP from being sent from the Asterisk in step 3? Does anyone know why the Asterisk sends RTP to remote host private IP? I would expect NAT=YES to correct this issue. Thanks, Peter ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 ---End of Original Message- ___ 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] bristuff patches and realtime mysql
Christoph wrote: On Tue, 2005-07-12 at 09:42 -0500, Matthew Boehm wrote: How do you expect me to possibly fix my module if you don't supply any compile errors? I don't use BRI so you will need to provide me alot of info. Which bri package did you install? Perhaps I can install that in a tmp dir and see what results I get. -Matthew Hi Matthew, I didn't want to spam the mailinglist with all those error messages generated by the compiler. I used the latest bri package from junghanns.net: http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC8h.tar.gz The package then downloads libpri-1.0.8, zaptel-1.0.8, cwain and asterisk-1.0.8 and does a lot of patching to the sources. This is then the error message that I get when trying to compile the asterisk-addons: pound:/usr/src/asterisk-addons# make all cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_saycountpl.o app_saycountpl.c cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o 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: In function `my_load_module': cdr_addon_mysql.c:268: warning: assignment makes pointer from integer without a cast cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 It would be really great if you could get it to work as I think it's an excellent addition for Asterisk. If you need more info just write me a message, Well, first off the error's you pasted above have nothing to do with res_config_mysql so I don't see where that comes into play. Interestingly enough, you are now the 3rd person to complain/write a subject line about res_config_mysql compile problems when infact there are no res_config_mysql compile problems stated in the email. Make sure you are using asterisk-addons from 1.0.8. and not HEAD which is what I am guessing you are doing. The very first line on the wiki for RealTime states that HEAD is required. There is no RealTime in STABLE. -Matthew ___ 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] NAT=YES
Mark Phillips wrote: Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf Anyone that tells you to use reinvite= is confused. The option does not exist (check the source code if you don't believe me). reinvite= is one of the many Asterisk Urban Myths. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] Any suggestions for an IP phone?
Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most important point. Thanks for any sugestion. -- Alexandre Leclerc ___ 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] IP Phone with Standard Power Ethernet
Kevin P. Fleming wrote: Eric Wieling aka ManxPower wrote: That's actually not correct. The IP600 supports PoE out of the box. The IP 30x and 50x support PoE with a special cable from Polycom. Bummer...I thought the built-in PoE chip was one of the few upgrades in the 300-301 and 500-501 paths... too bad, would have been a good thing to include. On the 301 and 501 I could be wrong. I thought the only change was more FLASH. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] meetme an customized menu
Hi, today i have taken a strong look at meetme.c what i am trying to accomplish is the following: it should be possible to access an menu from within the conference in order to perform special tasks, eg. to dial another number so that the called person is joined with the conderence. my first try was to use an agi-script for this, but as with agi enabled sip-channels (for which pseudo-zap-channels are created) doesn't send or get any audio. so, i decided to add this function in meetme.c in the standard menu you can press *1 to mute/unmute yourself and *2 (as admin) to lock/unlock the conf. what i have to do is to be able to let the user press *3. then there should be an announcement for the number. the user then dials the number and terminates with #. but i seems not to be able to recognize any dtmf-tone after *3 here a little test-code: case '3': /* Invite another Conferee */ menu_active = 0; ast_log(LOG_WARNING,Taste 3 gedrueckt - now one more for testing!); if (!ast_streamfile(chan, whatNumberToInvite, chan-language)) dtmf = ast_waitstream(chan, AST_DIGIT_ANY); else dtmf = -1; ast_log(LOG_WARNING,Something pressed ?? :%d \n,dtmf); break; My thougt was, that ast_waitstrem waits a certain time period and then return the dtmf-code. but somehow the code executes without pause between the two log-outputs. the original statement in the code, does indeed wait. any thougts ?? maybe someone has an idea how sip+agi+meetme could work ? thx in advance ... tobias wolf ___ 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] Pushing new firmware to Snom 190
On Tuesday 12 Jul 2005 15:51, Colin Anderson wrote: Thanks for replying. Frustrating, didn't work. Set it to update automatically, and made an HTML page consisting of: html pre bootloader: firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin /pre /html Added the HTML page to a webserver I control, and added it's URL to the Setting URL field. Rebooted. Didn't update. Also added http://www.snom.com/download/snom190-3.56m-SIP-j.bin to Software Update Firmware field, hit Load, nothing. System Information yields: Version-Code: snom190-SIP 3.44 -but- Firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin Phones that I have sucessfully updated when I am physically present to press the OK button yield: Version-Code: snom190-SIP 3.56m Firmware: http://www.snom.com/download/snom190-3.56m-SIP-j.bin Anyone got any other ideas? tia There are 2 problems here, the first is if you click on memory and the connection count is not 0, then you will be unable to reboot the phone, all you can do then is power cycle it. Secondly, to update the phone, you have to create 2 files, the first is entered into the Setting URL: and should at a minimum consist of something like: html pre firmware_status: http://81.187.187.52/snom190/snom190-firmware.htm /pre /html You then have to create a second file with reference to the above: HTML pre firmware: http://www.snom.com/download/share/snom190-3.60i-SIP-j.bin /pre /HTML It is a pathetically stupid way of doing things, but it is the Snom way. B ___ 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] Digium Wildcard TE110P IRQ problem
This worked for me:
before compile bristuff edit the file
wcte1xxp.c
near line 1526 initialize the array pci_device_id t1xxp_pci_tbl[]
this way:
static struct pci_device_id t1xxp_pci_tbl[] = {
{ 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x79be, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x793e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x791e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x799e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0 }
};
hope i helped you.
Regards
Accursio
[EMAIL PROTECTED] wrote:
Folks:
I'm trying to reach for help, I have a Digium Wildcard TE110P, on an E1,
the problem is every time the server reboots sometimes, and the system
does not recognize the card. I have on the Bios disabled the PNP OS and
the irq's are assigned manually, IRQ 9 for slot 3 (where I have installed
the card) bus still have the same problem.
Is there a way to fix or hardcode the irq for the card?
Thanks in advance for your help
JR
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RE: [Asterisk-Users] Cisco 7960 firmware upgrade promblems
And if it fails still, check for a buffer overrun on the configuration file SIPDefault.cnf, the lower firmware versions had less memory assigned for this file during the upgrade process. Caused me all sorts of problems :) Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Geoff Manning Sent: 12 July 2005 16:12 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7960 firmware upgrade promblems Sergio Chersovani wrote: I know it's hard to find out infos at the cisco site. Maybe you can open a TAC case Sergio I did find this info: http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20cisco%20 79xx comments_threshold=0comments_offset=0comments_sort_mode=commentDate_d esc comments_maxComments=10comments_parentId=353#threadId358 snip The two phones I purchased had Application Load ID (AKA: firmware) of P003AM30. This is their skinny protocol load. If you're trying to do sIP, you need a load that starts out POS.. You can not upgrade from P00 to P0S, you need to downgrade to P0S30203 to get it using POS firmware, then you can upgrade to the newer releases of the SIP firmware, with one extra thing to know. You do not need to step through every version of he firmware, you can jump versions of firmware, but what you encouter is the issue with their signed binaries (ie: *.sbn files) that they have converted to. If you have both a *.bin and a *.sbn file in the TFTP server root directory, it will default to loading the *.bin (ie: unsigned binary), which you do not want to do, since you need to convert over to signed binaries, in order to continue upgrading to get to the higher versions which only come signed. If you try to load higher version binaries that are not signed, the phone will fail to load and give an error as such (which I dont have the exact verbiage of). So, bottomline, go down to SIP 2.3, then go up to the first signed binary, then go to the final signed binary, then you ought to be there. /snip ___ 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 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] Any suggestions for an IP phone?
Alexandre Leclerc wrote: Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most important point. Thanks for any sugestion. You'll love the Polycoms, the IP600 is amazing. Get one on your desk and you won't want to let it go. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ 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] IP Phone with Standard Power Ethernet
Half of my 7960G phones work with standard POE, the other half work with the special rewiring. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kevin P. Fleming Sent: Tuesday, July 12, 2005 8:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Phone with Standard Power Ethernet Michael J. Tubby B.Sc (Hons) G8TIC wrote: Cisco 7940 and 7960 phones without the G (global) suffix used Cisco PoE and had their keys labled in local languages. The 7940G and 7960G Global phones are IEEE 802.3af PoE and have the keys engraved with icons and stick over labels. Are you sure about that? I have users with 7960G phones (icon buttons) that did not work without making cross-over patch cables from our PoE injectors. ___ 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 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] Cisco 7940/7960 interdigit timeout
Hello list, does anyone know how to change the interdigit timeout when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland ___ 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] Help Configuring TDM04B
Hi: I had an fxo card from Digitnetworks and it was working fine on my Asterisk box. I then replaced it with TDM04B. I changed the zaptel and zapata to include the four channels. When I run ztcfg, I get configuration errors: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Any suggestions beside the power plug I should try? Thanks Sell on Yahoo! Auctions no fees. Bid on great items. http://auctions.yahoo.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] RE: Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9
Good day Adam, I have about 30 Queues configured so at the risk of boring everyone I have included one of the lines; exten = _812108,1,Playback(nexus/wel-helpdesk-interwise) exten = _812108,2,SetCIDName(Client1) exten = _812108,3,Queue(Client1|Tt|||) exten = _812108,4,Playback(nexus/im-sorry) exten = _812108,5,Voicemail(1500) The _812108 is the DNIS number on the T1. I did have Tt configured in the queue. I followed your suggestion and changed this to; exten = _812108,1,Playback(nexus/wel-helpdesk-interwise) exten = _812108,2,SetCIDName(Client1) exten = _812108,3,Queue(Client1) exten = _812108,4,Playback(nexus/im-sorry) exten = _812108,5,Voicemail(1500) Same issue. I looked at the Agent's extension. It was configured as; ; Angela Holt exten = 0420,1,Dial(SIP/phone21,20,tr) exten = 0420,2,VoiceMail,u1021 exten = 0420,3,MusicOnHold(default) I changed this to; ; Angela Holt exten = 0420,1,Dial(SIP/phone21,20) exten = 0420,2,VoiceMail,u1021 exten = 0420,3,MusicOnHold(default) Removing the tr has done the trick. And the problem is gone. The agent can still transfer the call. Thanks for the idea. Warm Regards Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc I get greeted, put in a queue, given my position, the call goes through to my soft phone, and I accept the call, press #... I then get a message telling me that the system saying transfer? I see nothing on the CLI except the usual waiting for '#' to acknowledge Send the complete extensions.conf for the incoming call portion, and the agentcallbacklogin section. Also send the complete CLI from the call arriving into the PABX through to the call being sent to the agent. I suspect somewhere you are including the t or T option to the queue or dial which allows # to transfer a call. Of course, perhaps someone should check this, as we can't transfer a call until after we accept it... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Zaptel won't compile under Fedora Core 4
On Tue, 2005-07-12 at 11:04 +1000, Gonzalo Servat wrote: On 7/12/05, Eric Bullen [EMAIL PROTECTED] wrote: I hope someone can offer me some help with this. Basically, the current CVS version of Zaptel will not compile under Fedora Core 4. I have closely followed the directions in http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 using the versions given in the FC4 distro with no luck. Here's the output when I run make linux26. Any help would be great. TIA. [...snip...] In file included from /asterisk_source/zaptel/zaptel.c:40: /asterisk_source/zaptel/zconfig.h:10:27: error: linux/version.h: No such file or directory Try installing kernel-source and/or glibc-kernheaders RPMs from the FC4 CDs. The kernel package you need in FC4 is kernel-devel -- Telecomunicaciones Abiertas de Mexico Carlos Chavez Director de Tecnologia +52-55-91169161 Ext. 2001 signature.asc Description: This is a digitally signed message part ___ 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] IP Phone with Standard Power Ethernet
Interesting part is that I have none G and it worked with 3af On 7/12/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Michael J. Tubby B.Sc (Hons) G8TIC wrote: Cisco 7940 and 7960 phones without the G (global) suffix used Cisco PoE and had their keys labled in local languages. The 7940G and 7960G Global phones are IEEE 802.3af PoE and have the keys engraved with icons and stick over labels. Are you sure about that? I have users with 7960G phones (icon buttons) that did not work without making cross-over patch cables from our PoE injectors. ___ 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 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] NAT=YES
OK, so I have a nonexistant line in my settings. Why then when I remove it does my phone call fail? Rich Adamson wrote: FYI, there is no such thing as reinvite. Someone started using that in postings a long time ago and a few people keep posting it, but it doesn't exist. (Check /usr/src/astersik/configs/sip.conf.sample) Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf Mark Klint, Peter wrote: Good morning Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP. Configuration Info: I have all users in SIP.CONF configured with NAT=YES Asterisk has a public IP Remote host is behind a firewall with NAT When I sniff on the Asterisk public network, I see the following. 1. INVITE from remote host public IP to Asterisk public IP 2. 183 response from Asterisk public IP to remote host public IP 3. RTP from Asterisk public IP to the remote host private IP 4. RTP from remote host public IP to Asterisk public IP 5. RTP from Asterisk public IP to the remote host public IP Is there a way to prevent step 3 from happening? Or, is there a way to delay the invalid RTP from being sent from the Asterisk in step 3? Does anyone know why the Asterisk sends RTP to remote host private IP? I would expect NAT=YES to correct this issue. Thanks, Peter ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 ---End of Original Message- ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
Re: [Asterisk-Users] Any suggestions for an IP phone?
For multi-line without power over ethernet the snom 190 probably fits the bill and price point. I really like the 360 too, but is probably just over your price point. We purchased 50 at $225 mark. Many people are starting to mention the Grandstream 2000 as an option. For single line the uniden 200s have been good for us. Reliable good sound quality, but don't do the presence lights and multi-line stuff the snoms do. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Alexandre Leclerc [EMAIL PROTECTED]: Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most important point. Thanks for any sugestion. -- Alexandre Leclerc ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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: DTMF not sending properly via IAX
On 12 Jul 2005, at 15:05, Tony Mountifield wrote:In article [EMAIL PROTECTED],Mark Edwards [EMAIL PROTECTED] wrote: I'd be interested to understand where you read about the 'sequence numbers' issue. This sounds like it might relate to the problem I am experiencing. It's either that, or the DTMF is coming inband over the IAX channel... ;-) Well you could try doing a tcpdump to capture the incoming and outgoingpackets on the IAX port. Use the -w option with -s0, to write the wholeof each packet to the packet file.You could then use a recent version of Ethereal to read the packet fileand analyse it. Apparently, recent versions of Ethereal understand IAX.Ethereal is able to save RTP audio streams as audio files - it might beable to do the same with IAX audio streams, but I don't know. Having just spent an age getting my head around IAX2 sequence numbers,I might be able to help here.If one of you wants to mail me a packet trace I'll see if I can spot anything.Tim. http://www.westhawk.co.uk/ ___ 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] Cisco 7940/7960 interdigit timeout
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 12 Jul 2005, Roland Zagler wrote: Hello list, does anyone know how to change the interdigit timeout when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland The timeout is set in the dialplan.xml file with the Timeout tag. Like this: DIALTEMPLATE TEMPLATE MATCH=\*1.. Timeout=0 User=Phone/ TEMPLATE MATCH=* Timeout=5 User=Phone/ !-- Anything else -- /DIALTEMPLATE HTH - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD: 519961 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iQEVAwUBQtP5EUtP/KMNOfRbAQJkhQgAnsQz0fRx1t8vgFNnCq7w91eDMxisIKvH x6FoYwaWi2IldCnMx/+21wRotnrw5WJFOT/qVn7UmgPCA0sHtXA4gzf9uyNOZsTR oXYWXQ3hRm6s6XtU84BozQJnhLdavx10u5tI/KfG6IfNpXKIbukZDAxRxXuAW62b uL85pd2n2l7N4iFlih61Qj348UrVP65xU8ev0ri/N9VRC+A+/vQBRJwHY34ogm1O Ine2CaJ4sGZiwYJZE7WDSN8fh7/bklP2RJ8BDq8/AAC0g6Bs7gn5rh6jgmcWK0SC QcgZTN570xAo1xYcHCm2p8ZqwFdHpKPRdCENqFAPugV/rjEf7HV91w== =dV1P -END PGP SIGNATURE- ___ 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] Problem with DTFM and complex international setup
I don't think so. Your problem seems to do with your not being able to use an IAX client to transmit DTMF tones properly somehow. I am using a normal phone to connected to FWD which then connects to an Asteriskserver using IAX protocol. The point is that between the phone and the far Asterisk server, I guess that the tones are being sent as audio and not as inbound messages. So the far Asterisk server has to listen to the audio for the tones. On an unstable connection, it is sometimes missing a tone, or hearing a break in a tone and thinking that it is two identical tones, which results in a mis-interpreted number sequence at the other end. So I think the problems are different. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark EdwardsSent: 03 July 2005 02:20To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Problem with DTFM and complex international setup Do you think this might have an impact on http://bugs.digium.com/view.php?id=4631? Mark On 7/3/05, Mohit Muthanna [EMAIL PROTECTED] wrote: Right... that's the one. My mistake.On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote: I don't find this option in the Makefile. I find RADIO_RELAX which is something to do with radios and DTMF. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mohit Muthanna Sent: 01 July 2005 23:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with DTFM and complex international setup Try compiling Asterisk with RELAX_DTMF (See Makefile). Mohit. On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote: We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed Asterisk so that it is possible, using a PIN code, to dial out from Asterisk onto the local PSTN. This also works occasionally. Looking at the message from the Asterisk box it is clear that sometimes numbers are missed or repeated in the dial string. This I suspect is because Asterisk is listening to the DTMF tones but the signal is dropped; sometimes the drop means that a whole digit is dropped and sometimes is means that a digit is repeated. Does anyone know how I can fix this to make it more reliable (out-of-band DTMF?) or a better way to achieve a reliable setup? ___ 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 -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of people. Those who understand binary, and those who don't." ___ 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--Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of people. Those who understand binary, and thosewho don't."___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,Mark P. EdwardsFWD: 667917 ___ 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
