search for: hulber

Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango

Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault. Reverted to 1.6.0.6 and back to normal. ------------------ Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 EST 2009 x86_64 x86_64 x86_64 GNU/Linux Apr 3 11:49:56 asterisk kernel: asterisk[3780]: segfault at 00002ce1ac0537a8 rip 0000003e980715a8 rsp 00007fff5bf00c30 error 4 Apr 3 11:50:00 asterisk kernel: asterisk[3828]: segfault at 0000000004000000...

...ension '0114312297136' rejected because extension not found. [May 15 12:20:29] NOTICE[3420]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '8104312297134' rejected because extension not found. I've not seen them before until recently. -- MARK. Hulber Technologies asterisk-admin at hulber.com Read my blog : http://mark.hulber.com Follow @hulber on Twitter: http://twitter.com/hulber

Hi all, I'm new to this list, so forgive me if I'm not supposed to ask this: I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there any way to use TLS with this phone<--->asterisk (v 1.6.0.9)? It is said that is supports TLS/SRTP but I don't see any of these options in the configuration file or the admin (advanced) SIP conf panel. Am I missing something? Thnx

I started to have a problem today that all my calls through voipjet result in just timing out after my assigned timeout period. I tried multiple of their servers with the same problem. Anyone else having a problem? I am running: Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulber.com on a i686 running Linux on 2006-05-03 14:14:07 UTC I can connect with other IAX providers. MARK.

Greetings, I'm having a heck of a time with one way audio on a SPA2012. It's public IP connected directly to cable modem. One line configured. Asterisk is multihomed Public IP outside / Private Inside. Extensions inside network are can't hear audio from phone outside connected via the spa-2012. Outside can here audio from inside the network. Ring works both ways. I've

[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

I want to use a voip speaker phone as a room monitor. Requirements: A phone that I can set to auto answer in speaker mode. A phone with a good speaker phone. Ability to make the audio one way. I want to monitor the room but not have my voice heard in the room. Yes, the mute button can accomplish this also. I have been using the SPA942's around the house (the speaker is just ok but

1.6.1 svn 190575: CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" menuselect make[1]: Entering directory `/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect' gcc -m64 -march=native -mtune=native -floop-interchange -floop-strip-mine -floop-block -c -o

Greetings! I'm hoping someone can help me with what should be the most basic of problems. Essentially, I want to have certain calls on an Asterisk 1.2.25 (Yes I know its old, upgrade, etc... its on my roadmap) install go out a couple of analog lines and all other calls go out a PRI. The analog lines are setup in Zaptel group 1 and the PRI channels are in Zaptel group 0. Here is my relevant

It looks like there's a problem with the location or naming of the Extra SLN16 sounds: --14:11:43-- http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz Resolving downloads.digium.com... 76.164.171.232 Connecting to downloads.digium.com|76.164.171.232|:80... connected. HTTP request sent, awaiting response... 301 Moved

Hi, I am using Asterisk 1.4.25. I have one Broadvoice account. I Integrated this broadvoice account with Asterisk Server. I am Able to Make calls but cannot recieve calls. In Incoming calls, call lands to SIP extension, as I attend the call....It gets hungup......... If i dont transfer this call to extension or I play any file then it works OK. But as I transfer it to SIP Extension it get

-------- Original Message -------- Subject: Re: [Asterisk-Users] BroadVoice configuration changes for Outbound Date: Sun, 06 Mar 2005 19:11:22 -0500 From: MF Hulber <mark@hulber.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>, dan@mirrorlynx.com References: <200503060703.XAA12457@comand.net> <422AB1E3.6020803@cox.net> <1299.192.168.1.101.1110097439.squirrel@lexon.ws> <111...

Thanks Mark will try that out! -----Original Message----- From: MF Hulber [mailto:asterisk-admin@hulber.com] Sent: 22 March 2005 05:25 To: Reuben Grech Subject: [info] [Asterisk-Users] :: BIOS Motherboard Settings :: I have the same motherboard. I put the card in the 2nd slot from the bottom. In this slot, if you look at the manual, it will possibly be in conflict wi...

Previously, when I wanted to forward to incoming callerid when I forwarded a call to another number I had to set the callerid on the outgoing call to be that of the incoming number. So today I do this: exten => s,n,Set(CALLERID(name)=${CALLERIDNAME}) because I want the outgoing callerid that I forward to not be the normal callerid of the local extension but I want to forward the incoming

I have a new Polycom Spectralink 8002 and am having trouble with the configuration or the unit but I can't see what's wrong. The unit does not seem to even attempt to register with the Asterisk proxy but I can make calls to it. I have viewed the syslog from the device which it will actually write to the asterisk server so I know it can be reached. I have also run a sip debug and

How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/f9612f97/attachment.htm

Although I'm convinced that Broadvoice doesn't have the most stable of ping times, it seems like I get ping results that are approximately the ping time +2000ms at times. Has anyone experienced this problem with qualify on a SIP connection before? So here, was the ping 20ms or 2020ms as reported? Aug 22 06:39:49 NOTICE[6964]: chan_sip.c:8481 handle_response_peerpoke: Peer

...production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: &g...

Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat: