Re: [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider

2008-03-22 Thread Joshua Kinard
-Original Message- From: [EMAIL PROTECTED] On Behalf Of Ignacio Ortega A. > Let me tell you something i own 200 seats call center, besides have an IT > company who develops applications based on asterisk, we are not kid just > playing to get some money... so i move millions of min i a

Re: [asterisk-users] Newbie: Two problems with Asterisk Config, Please Help

2008-03-22 Thread Pete Kay
Hi, I can get the message recorded and played correctly with wengo, but not with zoiper. Is there any codec setting that I should fixed and how to fixed it? On Fri, Mar 21, 2008 at 9:26 PM, Steve Totaro < [EMAIL PROTECTED]> wrote: > Probably a codec issue. SIP debug while making a call would be

[asterisk-users] how to detect redirect fax call

2008-03-22 Thread Pete Kay
Hi, I want to try detecting if a call is a fax from Zap/1 channel and if it is, forward it to a fax number. How to do it? I have iaxmodem and Hylax working, but it can only receive, but not redirect the fax call. Also, I have read that Asteris has a tool call rxFax. Could someone help me to unde

[asterisk-users] Anyone used Siemens SIP/Dect phones?

2008-03-22 Thread Alan Lord
Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the "soon-to-become-available-in-the-uk" S685IP. Both systems have great feature sets and, on-paper at least, look to be the bee's knees.

Re: [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)

2008-03-22 Thread Faraz Khan
Did you manage to upload those changes? Some of your schema/ldif files were deleted by the bug admin. You might want to upload them at voip-info Furthermore, the multi_ldap call is broken in res_config_ldap.c - I even started a bounty on it but looks like few people are interested and/or b

Re: [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)

2008-03-22 Thread Faraz Khan
Just checked 1.6.0beta4 - the res_ldap.conf file still has PBX* attributes - which I'm guessing would be confusing to any new user. the schema file looks file though, the missing voicemail/queue part is what we have added. Quoting Faraz Khan <[EMAIL PROTECTED]>: > Did you manage to upload t

Re: [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider

2008-03-22 Thread Ignacio Ortega A.
Our facilities are in USA in albany NY with USA personel taking care of it, and please dont juge an entire country for one or two you have know before, my company is honest and legit and again we are very sorry for post this message here, what else you want me to say??? On Sat, Mar 22, 2008 at 3:0

Re: [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider

2008-03-22 Thread Outback Dingo
Ok well seems there is at least one thing legit, and that is there does exist a city Albany in NY :) ok nuff joking, first if you want to really be taken seriously, fix up your web site, you need something a "bit" more "professional, and you also need a lot more company info on it. Im sorry but, it

Re: [asterisk-users] BLF and Snom phones

2008-03-22 Thread Philipp Kempgen
Loic Didelot schrieb: > I am having some troubles with Snom phones and maybe someone can help > me. > > Let me say this: BLF and pickup works great with Polycomes and > Grandstream etc... So I think my problem might not be Asterisk related > but I am not 100% sure. > > The snom phones subscribe

Re: [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider

2008-03-22 Thread Ignacio Ortega A.
suggestion accepted !!! we already change the email address and added a 1800 number that will be available from Monday for direct support and questions to be honest we focuced more in the billing platform an the provisioning of services maybe we didn`t take care much about the page itself but thank

[asterisk-users] Problem: Digium TDM400 with XOptionsFlex - Line Busy

2008-03-22 Thread Thomas Klettke
Perhaps someone with more experience can help me solve this puzzle: Asterisk 1.4.18 on a Dell PowerEdge SC440, CentOS 5, 2.6.18-53.1.14.el5, Digium TDM400 with 3FXO(1-3) and 1FXS(4) Phone service is provided by XO Communications via their XOptionsFlex service (5 analog lines, 3 of which are used

Re: [asterisk-users] Anyone used Siemens SIP/Dect phones?

2008-03-22 Thread Gordon Henderson
On Sat, 22 Mar 2008, Alan Lord wrote: > Hi all, > > I am close to purchasing some new DECT phones for our home office here > in the UK. > > We use Asterisk and I am sorely tempted by the Siemens C475IP or the > "soon-to-become-available-in-the-uk" S685IP. > > Both systems have great feature sets a

Re: [asterisk-users] Anyone used Siemens SIP/Dect phones?

2008-03-22 Thread Michael Graves
On Sat, 22 Mar 2008 16:50:39 + (GMT), Gordon Henderson wrote: >I'd look at the Snom M3's but they're more expensive that my customers >will pay )-: And had a report of them being unstable too )-: > >So for the time being I'm sticking to ATA's and cheap analogue DECT phones >for those custome

Re: [asterisk-users] Calls to sip extensions not defined

2008-03-22 Thread R. B.
Thanks Steve, your solution works but I am looking for is something more general. The example I posted is a simplified one and on the real one I am using extensions on the 5XXX scenario so I can have 5000 to 5999 range. You answer for just 10 extension is great but for a 1000 and when not all exten

Re: [asterisk-users] Problem: Digium TDM400 with XOptionsFlex - Line Busy

2008-03-22 Thread John Novack
Thomas Klettke wrote: > Perhaps someone with more experience can help me solve this puzzle: > > Asterisk 1.4.18 on a Dell PowerEdge SC440, CentOS 5, 2.6.18-53.1.14.el5, > Digium TDM400 with 3FXO(1-3) and 1FXS(4) > > Phone service is provided by XO Communications via their XOptionsFlex > service (

Re: [asterisk-users] Anyone used Siemens SIP/Dect phones?

2008-03-22 Thread Alan Lord
Gordon Henderson wrote: >> >> Anyone got any skeletons on them? > > I've deployed a number of Siemens C460IP's. > > They're really good and coverage is excellent, but for one thing: They > base stations lose registration with the asterisk box after some time and > need rebooting. I've read tha

Re: [asterisk-users] Anyone used Siemens SIP/Dect phones?

2008-03-22 Thread Gordon Henderson
On Sat, 22 Mar 2008, Alan Lord wrote: > Gordon Henderson wrote: > >>> >>> Anyone got any skeletons on them? >> >> I've deployed a number of Siemens C460IP's. >> >> They're really good and coverage is excellent, but for one thing: They >> base stations lose registration with the asterisk box after

Re: [asterisk-users] Hardphone SIP phone costs

2008-03-22 Thread Norman W. Franke
On Mar 21, 2008, at 11:48 AM, [EMAIL PROTECTED] wrote: > I'm getting the impression that the telcos in the US are basically > shafting you because of the monopoly they have. More intersted in > keeping > themselves happy than their customers. I think it's nice I have a > choice > of 5 major m

Re: [asterisk-users] Calls to sip extensions not defined

2008-03-22 Thread Steve Edwards
On Sat, 22 Mar 2008, R. B. wrote: > [default] > exten = _1[1-3],1, dial(sip/${EXTEN},10) > exten = _1[1-3],n, voicemail([EMAIL PROTECTED],u) > exten = _1[1-3],n, hangup > exten = i,1, playback(pbx-invalid) > exten = i,n, hangup > Thanks Steve, your solution works but I am looking for is som

[asterisk-users] G723 on asterisk 1.4.1

2008-03-22 Thread wassim darwish
Hi: How to install and set up my asterisk server with G723 codec to send and receive calls using it. Thanks in advance; Wassim _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&f

Re: [asterisk-users] Which command line is used to send emails to notify incoming voicemail ?

2008-03-22 Thread Robert Lister
On Fri, Mar 21, 2008 at 12:21:09PM +0100, Olivier wrote: > > In Asterisk full log, I can see > Mar 20 14:36:41 DEBUG[29025] app_voicemail.c: Sent mail to > [EMAIL PROTECTED] command '/usr/sbin/sendmail -t' > > But when I type "/usr/sbin/sendmail [EMAIL PROTECTED]" I can't see the same > log lines

Re: [asterisk-users] Anyone used Siemens SIP/Dect phones?

2008-03-22 Thread Robert Lister
On Sat, Mar 22, 2008 at 09:08:43AM +, Alan Lord wrote: > Hi all, > > I am close to purchasing some new DECT phones for our home office here > in the UK. > > We use Asterisk and I am sorely tempted by the Siemens C475IP or the > "soon-to-become-available-in-the-uk" S685IP. Have been using t

Re: [asterisk-users] Anyone used Siemens SIP/Dect phones?

2008-03-22 Thread Chris Bagnall
> They're really good and coverage is excellent, but for one thing: They > base stations lose registration with the asterisk box after some time and > need rebooting. I've not played with the C460s, but have deployed plenty of S450s, and they did indeed exhibit this behaviour in earlier firmware

Re: [asterisk-users] Anyone used Siemens SIP/Dect phones?

2008-03-22 Thread Chris Bagnall
Can’t comment on the C460, but the S450 definitely doesn't have these issues: > - No SIP call transfer feature (that I can find) Hit "ext call" during a call, create a new call, then you can SIP transfer between them. > - Base station supports multiple phones, but you can only register each >

Re: [asterisk-users] G723 on asterisk 1.4.1

2008-03-22 Thread Martin
Download an appropriate binary from http://asterisk.hosting.lv/ and just drop into /usr/lib/asterisk/modules/ add allow=g723 to your sip.conf as necessary and restart asterisk... Im only not sure how legal is this, you will probably need to obtain licenses for all concurent channels... Martin ---

Re: [asterisk-users] Calls to sip extensions not defined

2008-03-22 Thread Stefan Schmidt
hello, why do you build a realtime configuration loading the sip users and extensions from a database? if you want to use the username as extension it would be quit simple looking like this: exten => _X.,1,Select user from DB which has ${EXTEN} exten => _X.n,GotoIf(Uservar = ""?nouser) exten =

[asterisk-users] voicemail and needed language to be selected

2008-03-22 Thread bilal ghayyad
Hi List; I need to include Arabic Language for the voicemail, anyone can advise me for the needed steps to proceed? I can record the messages in arabic, but how to let the voicemail selected the arabic recorded voice messages instead of the english? Any advise? Regards Bilal __

Re: [asterisk-users] voicemail and needed language to be selected

2008-03-22 Thread Alex Balashov
Replace the vm-* recordings in /var/lib/asterisk/sounds? bilal ghayyad wrote: > Hi List; > > I need to include Arabic Language for the voicemail, > anyone can advise me for the needed steps to proceed? > > I can record the messages in arabic, but how to let > the voicemail selected the arabic re