similar to: VSP authentication to incorrect context

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

Hi All, I'm having problems getting results from a PHP file. This is what the CLI is showing. -- Executing [111 at internal:1] AGI("Console/dsp", "GoTalk.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/GoTalk.php [Dec 14 11:57:25] ERROR[20260]: utils.c:1019 ast_carefulwrite: write() returned error: Broken pipe If I run the PHP file from

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have been

=20 arp | grep "192.168.0.1" =20 substituting the IP address of the SIP device. =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client MAC address. =20 hello, is

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to compile Samba on an Intergraph 6750 server running Clix. These are my results.................... S_GIS/usr/gis/doug/samba/samba-1.9.18p3/source>make Using CFLAGS = -O -DSMBLOGFILE="/usr/local/samba/var/log.smb" -DNMBLOGFILE="/us Using LIBS = -lbsd -lc_s Compiling mem_man/mem_man.c Compiling rpc_pipes/srvparse.c Compiling rpc_pipes/smbparse.c Compiling

What does this mean for the SAMBA community? http://www.intergraph.com/press98/as050698.stm WINDOWS NT SERVICES FOR UNIX ADD-ON PACK NOW IN DEVELOPMENT http://www.microsoft.com/corpinfo/press/1998/May98/ntunixpr.htm The Windows NT Services for UNIX Add-On Pack will make it easier to integrate Windows NT Workstation 4.0 and Windows NT Server 4.0 into existing UNIX environments.

All, I have a Polycom 650 phone, when turned on displays "Checking application". Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/daadda2a/attachment.htm

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J.

I have a brand new TE120P card that I have installed and asterisk is not starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown signalling method 'pri_cpe' It seems it does not matter what I change the vaule for signalling= to, it always returns it as invalid. I have tried the config from my other 2 servers running TE110P cards and the config from AusTechPartniships

Hi guys, I currently have about 10 Asterisk servers scattered around the place each hosting their own dynamic conference centre. Is there any way that when people join these conference centres on each server that somehow Asterisk bridges the conference centres on each server to form one large conference? Many Thanks David. -------------- next part -------------- An HTML attachment was

Hi All, I'm stuck and need some help. I have installed the Asterisk CDR Analyser Version 2.0.1. It mostly works except for the CDR Report. I get the following error even though it lists the CDR details. Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day, sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE UNIX_TIMESTAMP(calldate) >=

I'm sure this might not be the correct place to ask and I have done a Google but I can't seem to find anything that says there is a VSP that will work with * in the Ukraine. I have a friend that lives in Kiev and basically want a phone number there to be able to talk to him and have him call me. If anyone has any information on it and they are willing to share please advise.

Hi All, I think I have missed something as I am resisted with 4 VSPs and I can not dial in using any one of them using the corresponding VoIP numbers assigned with the VSP. I can make outbound calls to another VoIP number to the same provider. The weird thing is that I have a DID with a VSP and I have that working fine but try using the associate VoIP number and nothing happens. When

Hi All, I seem to be having a problem with all my VSPs. When I am registering with them I don't seem to be passing my port number. This problem causes other users the inability to call my VoIP number with the VSP. My VSP showed me what they are seeing. I have changed my useragent to be: Linksys/SPA941-4.1.15 Linksys/SPA941-4.1.15 Contact sip:1234321234@aa.bb.cc.dd with no

The Telco in Canada is been real painful. I was wondering if anyone has installed a Digium TE1X0P card in Canada and if their Telco was so difficult. The Telco will not provide us a service until they see a FCC or DOC number for the equipment ware are connecting to their service. If have found "FCC Part 68, ANSI/ITA-968-A, Including Amendment A1 and A2 Industry Canada CS-03"

Greg, Chris, and Jay, Thanks! You've given me plenty of info to digest. I really appreciate the responses. Apologies if my list manners aren't up to snuff! Thanks again, Jason