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...nf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN}) exten=>_y.,1,Dial(IAX2/${EXTEN}) exten=>_a.,1,Dial(IAX2/${EXTEN}) below is my Asterisk console logs which i see after making call. Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No registration for peer 'aliadvcommnet' (from 203.99.57.80) Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No registration for peer 'aliadvcommnet' (from 203.99.57.80) Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket_read: R...

...er context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN}) exten=>_y.,1,Dial(IAX2/${EXTEN}) exten=>_a.,1,Dial(IAX2/${EXTEN}) below is my Asterisk console logs which i see after making call. Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No registration for peer 'aliadvcommnet' (from 203.99.57.80) Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No registration for peer 'aliadvcommnet' (from 203.99.57.80) Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket...

...er context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN}) exten=>_y.,1,Dial(IAX2/${EXTEN}) exten=>_a.,1,Dial(IAX2/${EXTEN}) below is my Asterisk console logs which i see after making call. Mar 28 03:25:43 NOTICE[2855]: chan_iax2.c:5144 register_verify: No registration for peer 'aliadvcommnet' (from 203.99.57.80) Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:5144 register_verify: No registration for peer 'aliadvcommnet' (from 203.99.57.80) Mar 28 03:25:55 NOTICE[2855]: chan_iax2.c:6910 socket...

http://grandstream.com/BETATEST/HT488_496_386/

...; May 26 08:06:33 DEBUG[27570] manager.c: Manager received command 'Command' May 26 08:06:33 DEBUG[27570] manager.c: Manager received command 'Command' May 26 08:06:34 VERBOSE[28367] logger.c: == Connect attempt from '10.34.9.135' unable to authenticate May 26 08:06:46 DEBUG[2855] chan_sip.c: Stopping retransmission on '68e15a667d45ae9a1f07434c2eaa6d53@10.34.9.206' of Request 102: Match Found May 26 08:06:59 DEBUG[2855] chan_sip.c: Auto destroying call 'CB895D7B65644EAE9105A5624E823ED8@10.34.9.206' May 26 08:06:59 DEBUG[2855] chan_sip.c: Auto destroying call...

...710, 3116, NA, NA, NA, NA, NA, NA, NA, NA, NA, NA, 2993, 2993, 2965, 2852, 2701, 2606, 2663, 2663, 2663, 2701, 2587, 2616, 2738, 2757, 2757, 2757, 2720, 2701, 2833, 2767, 3840, 3840, 3840, 3840, 4150, 3950, 3500, 3600, 3600, 3600, 3800, 3700, 3600, NA, 3395, 3395, 3395, 3400, 3300, 3135, 3070, 2855, 2855, 2855, 2865, 2800, 2810, 2900, 2845, 2845, 2845, 2915, 3030, 3030, 3010, 4850, 4850, 4850, 4850, 5550, 6350, 6800, 6100, 6100, 6100, 6600, 6550, 6000, 5800, 5650, 5650, 5650, 5700, 5800, 5600, 5000, 5000, 5000, 5000, 5300, 5200, 5300, 5350, 5750, 5750, 5750, 6600, 6400, 6700, 6700, 23708,...

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

...H Per-Connection Daemon ([::1]:60865). systemd[1]: Starting OpenSSH Per-Connection Daemon ([::1]:60865)... systemd[1]: Started OpenSSH Per-Connection Daemon (127.0.0.1:41286). systemd[1]: Starting OpenSSH Per-Connection Daemon (127.0.0.1:41286)... sshd[2854]: Connection closed by ::1 [preauth] sshd[2855]: fatal: ssh_dispatch_run_fatal: Connection reset by peer [preauth] Looks like this happens if we have two incoming connection (::1 and 127.0.0.1 are checked) at the some time. Why does this happen? Who's fault is it? As these are TCP connections I would expect it is not a problem to know what...

...o my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote server. Thanks, -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856

...t configure:2721: $? = 1 configure:2744: checking for C compiler default output file name configure:2771: gcc conftest.c >&5 configure:2774: $? = 0 configure:2820: result: a.out configure:2825: checking whether the C compiler works configure:2835: ./a.out configure:2838: $? = 0 configure:2855: result: yes A difference may be that I build for x86_64. Regards, John

On 24/11/2007, at 9:33 AM, John van Schie wrote: > Antony Blakey wrote: >> On 24/11/2007, at 12:01 AM, Michael T. Richter wrote: >> >>> Does anybody have a working version of LLVM 2.1+ (ideally in a .deb >>> file) that they're willing to share? I'm having some weird problems >>> trying to get a working version on my 7.10 machine. >>

Antony Blakey wrote: > On 24/11/2007, at 12:01 AM, Michael T. Richter wrote: > >> Does anybody have a working version of LLVM 2.1+ (ideally in a .deb >> file) that they're willing to share? I'm having some weird problems >> trying to get a working version on my 7.10 machine. > > Or even better, does anyone have a recipe for building on Gutsy? I've just

Not sure what's up with the mailing list here. For some reason mails are not coming through. Try again... I am trying to link an asterisk box to my provider's asterisk server via SIP. (I know I could use IAX, but the provider does not allow that, so I can't). When an inbound call happens I get this: Jan 31 13:09:14 NOTICE[3716] chan_sip.c: Failed to authenticate user

Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

...rectly following the Win->Win sync shows 104 directories and 52000 files copied. In my world they should be identical. Total Copied Skipped Mismatch FAILED Extras Dirs : 5621 104 5517 0 0 6 Files : 54888 52033 2855 0 0 19 Bytes : 20.645 g 19.767 g 899.69 m 0 0 21.91 m Times : 0:10:00 0:07:29 0:00:00 0:02:30 The Samba settings are [global] bind interfaces only = Yes client min protocol = SMB2 dedicated keytab file = /etc/krb5.keyta...

...t;robocopy z:\ d:\temp test.bin Total Copied Skipped Mismatch FAILED Extras Bytes : 1.895 g 1.895 g 0 0 0 0 Times : 0:00:40 0:00:40 0:00:00 0:00:00 Speed : 49902 KBytes/sec. Speed : 2855 MBytes/min.

...2017 at 22:17, Adi Pircalabu <adi at ddns.com.au> wrote: > May I chip in here? In our environment we're randomly seeing: > > Jan 17 23:40:14 xen01 kernel: ixgbe 0000:04:00.1 eth6: Detected Tx Unit > Hang > Someone in this thread: https://sourceforge.net/p/e1000/bugs/530/#2855 reported that *"With these kernels I was only able to work around the issue by disabling tx-checksumming offload with ethtool."* However, that was reported for Kernels 4.2.6 / 4.2.8 / 4.4.8 and 4.4.10. I just thought it could be something you could rule out and hence mentioned it: etht...

Hi I have set up an Asterisk System with One XLite Phone and when i call the trunk line or receive calls via a trunk line (FXO generic X100P) i'm getting one way Voice. I can hear the called party - but they cannot hear me... Any ideas - is t a NAT issue or is it something to do with the generic X100P card. How does one sort this problem -- Mark D'Cruz D'Cruz Consulting

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on