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...e a rhtml file that goes through the request.params. One of my test generates that file, but rspec pops an error saying : undefined method ''params'' for ... I''m not sure what to do with this, since request.params actually works with rails. Any idea? Thank you Olivier Dupuis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/rspec-users/attachments/20080411/e61f03cf/attachment.html

...''E''" do @language = "e" render "/search/index" response.should have_tag("a[href=?]", "/search/index/f") end I get the error message saying that no tag was found. Any help would be appreciated Thank you in advance Olivier Dupuis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/rspec-users/attachments/20071129/72a905bc/attachment.html

...#39;', :id => ''f'' response.cookies["thothle_language"].should equal(''f'') end The error I get is the following: ...expected "f", got ["f"] (using .equal?) Any help would be appreciated. Thank you Olivier Dupuis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/rspec-users/attachments/20071128/63f7e63b/attachment.html

Okay, I've upgrade samba, now I use samba3.schema who is with my suse 9.2 So I delete all in /var/lib/ldap and in /var/lib/samba Redo smb-populate blablabla (from the howto http://samba.idealx.org/smbldap-howto.en.html) So now when i would like to join my Samba domain : [2005/03/21 15:45:51, 2] auth/auth.c:check_ntlm_password(312) check_ntlm_password: Authentication for user

I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x "restart gracefully" However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD

We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the "only telco's get documentation" crap) Does anyone have a suggestion? Thanks, MD -------------- next

...uding being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules / virtual user tables). They also include error logs, parameter checking, etc. Let me know if you want them Michelle Dupuis Technical Support Specialist Oxford Consulting Group Ltd. Making IT work for your business... T: (519) 672-8238 E: <mailto:support@ocg.ca> support@ocg.ca W: <http://www.ocg.ca/> www.ocg.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists....

I have a new asterisk system with a T1 card. It appears that running "ztcfg -vv " is required in order for asterisk to start properly. Is this correct? Are people adding this command to the asterisk startup script? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

We're creating a SIP gateway for a client that will take one leg of a call in via SIP, and out the other side via H.323. To minimize load on the gateway, we would like to have the RTP stream bypass the gatewayy altogether (directrtp/reinvite). Is this possible with these to protocols? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm looking for recommendations for standalond PRI to SIP converters. (Needs to be outside the asterisk box - so a PCIe card won't do) I've used redfone but this project doesn't need the redundancy features... Thanks!

Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)....also hoping for something more current. I suspect RH5 and RH6 are most popular...but I'm looking for facts -------------- next part -------------- An HTML attachment was scrubbed... URL:

Over a year ago I saw a discussion about a standalone device which converted a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device is? (I'm looking for a standalone device - not a PCI card). Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070907/7060b23e/attachment.htm

I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071026/9cea5e74/attachment.htm

...mb_reh=0 smb_err=0 smb_flg=136 smb_flg2=49153 smb_tid=0 smb_pid=5220 smb_uid=0 smb_mid=1 smt_wct=17 smb_vwv[ 0]= 8 (0x8) [...] 5220: session setup failed: ERRDOS - ERRnoaccess (Access denied.) So, does mount.smbfs support NTLMv2 ? Is there some special option/compile flag to use ? TIA Pascal Dupuis P.S.: I am (no more) subscribed to this list, could you please CC replies ? -- Dr. ir. Pascal Dupuis K. U. Leuven, ESAT/ELECTA (formerly ELEN): http://www.esat.kuleuven.ac.be/ Kasteelpark Arenberg, 10; B-3001 Leuven-Heverlee, Belgium Tel. +32-16-32 10 21 -- Fax +32-16-32 19 85

Has anyone written an app that monitors SIP/IAX registration attempts? A couple of clients are being flooded with SIP registrations (but the source IP changes every few hours so IPtables won't do).. I would think that any attempt to reg 5 times with a bad password should cause a 5 minute timeout until reg is considered again. Has anyone written such an app? The name app_hackblock is my

Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150627/6774c750/attachment.html>

I found some great pieces of script on the internet that I've combined to allow Asterisk to send voicemails as an MP3 file, and encode the sender name and number as well as message number as tags into the MP3 file. I even include a "cover art" image which has our company logo and PBX symbol in it. Mobile phone users love it, and Android phones can now play the attachments (without