similar to: Analog phone not ringing

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm just learning about VoIP and Asterisk. I've got a developers kit on its way and I've managed to get hold of a couple of cheap Cisco VIP 30 phones. I've trawled the web and found a few snippets of information on these phones but I still can't get them to work. Does anyone have any config files or any idea on how (if I can)

Hello again, I'm not attempting an install of Samba 3.0.4 from source. I want to specifically compile in ads and winbindd support. So, I already compiled and installed OpenLDAP 2.2.13. Now the configure script is hung on krb5 dependancy: checking for krb5.h... no configure: error: Active Directory cannot be supported without krb5.h So I downloaded and attempted compile of krb5 1.3.4, but

I have put my phpconfig stuff out into the Digium CVS tree. Project name is phpconfig. see it at http://rads.netcom.utah.edu/phpconfig/phpconfig.php Lemme know if you have any patches or add on's are welcome Dave Packham aka p0lar

I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v

Hi all, can somebody explain me why i can't hear a ringing tone (alerting) if i'am going to connect to my destination end point? Is it basically so that i have to configure like: exten => xxx,1,Dial,ChanTec/number|timout|r Is it really nessesary to use the "r" option everytime if i want to indicate a ringing tone? This suggest a wrong call flow for the user ... Thanks for

Here's a bit of a twist to the common UK Caller ID question... (Which I've got working nicely thanks to some slight changes in Jonathan McHarg's scripts off the asterisk-dev mailing list, and a Pace modem from ebay!) Can a standard BT phone that supports CID (Such as a BT Decor 310) pick up the CID information that asterisk passes out to analog lines or would I have to get an

I'm looking at using a modem to provide caller ID info on my NTL line following the steps in the article posted by Darren Poulson: http://www.22balmoralroad.net/modules.php?name=Sections&op=viewarticle&artid=1 I was wondering if anybody had experience of using any modems (can be pci/isa/external) with an ntl line (in an ex. Cable & Wireless area if it makes a difference) and

Hi, I've got a recent copy of CVS (09/08/04) with the UK CID patch on and working, but I'm having trouble receiving SMS messages. I can send messages without any problems but whenever an incoming message is detected, nothing is left in the /var/spool/asterisk/sms directory. Can anyone help? Thanks, Darren. extensions.conf: [default] exten = s/08005875290,1,SMS(testq,a) exten =

Just did a complete cvs checkout (Sep 4, 9:30am cdt). Seems any attempt to dial out via a tdm04b (fxo) now fails. Does not seize the pstn line. CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo inbound-bus-x10 en default 1 inbound-home en default 2

Hello all, Forgive me as this is probably a common question. However, I could not find an answer while searching. I have a few Linux servers running Samba 2.2.x that are hosting open shares with guest read/write access. I would like to have these become a member of the active directory and allow pass-thru authentication and authorizations to the shares hosted. I find where I can join the

Hi all My * implementation is working brilliantly with only one small fault left to kill. I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the pstn network; if I set my codec to GSM everything works great - no pauses but quality is a bit poor. If it set the codec to alaw (I think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call.

Hi everyone I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI analog problem. (see [Asterisk-Users] Multiple DDI & Hunting on Analog Lines (UK) for details) Does anyone on the list have any recent comments on reliability etc? I would really appricated some positive and negative comments. Cheers Matt

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

Dear R-experts, Sorry if I've overlooked a simple solution here. I have calculated a proportion of the number of observations which meet a criteria, applied to five years of data. How can I break down this proportion statistic for each year? For example (data in zoo format): open high low close hc lc 2004-12-29 4135 4135 4106 4116 8 -21 2004-12-30 4120 4131

hi how can I find current codec from an AGI scipt? roy

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Hello, I have found a "bug", I think in the way TDM400P cards handle FXO interface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO

For those who missed it, there's a new release of aoTuV (beta 4.5). According to Aoyumi-san, there's some new tuning for audio below quality set 3 (-q3), but I have not tested it myself yet. I wonder when Monty will actually check the new versions of aoTuV and merge them with main trunk. Latest tests on HA show the latest tunings of Aoyumi-san have matured Vorbis a lot, and

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Ahh. Seen that before! That suggests to me that you don't have your sip.conf records setup right. What's your sip.conf look like? On 15-05-28 04:51 PM, Luca Bertoncello wrote: > Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > >> The phone you gave your wife is really old. Are you sure it supports SIP >> OPTIONS? Can you make a call in or out to it?