Re: [asterisk-users] Friday Feb 29th Leap Year Special wih Aastra

On Fri, Feb 29, 2008 at 2:08 AM, Rob Hillis [EMAIL PROTECTED] wrote: If anyone has managed to compile and run Asterisk on a server from this particular era, I'd love to know about it. :) Reports on asterisk and SIP from Roman times are a little sketchy. However, in about 110 BC, Claudius

[asterisk-users] Openser balancing Asterisk

Hi , i've a problem with the dispatcher module of Openser, for the load balancing for asterisk The schema of the network is this : Firewall (public ip: 199.199.199.199:5060) |

Re: [asterisk-users] quickfix for building zaptel with 2.6.24?

On Thu, Feb 28, 2008 at 11:10:37AM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: zenon:~# module-assistant -t build zaptel make[3]: Entering directory `/usr/src/linux-2.6.24.3' scripts/Makefile.build:46: *** CFLAGS was changed in

[asterisk-users] Cisco 7965g and asterisk

Hi, We've just bought a new cisco 7965g and web are trying to connect it to asterisk. I've bought smartnet and downloaded cmterm-7945_7965-sccp.8-3-3SR2.exe cmterm-7945_7965-sip.8-3-3SR2.zip The zip file contains: Archive: cmterm-7945_7965-sip.8-3-3SR2.zip Length Date TimeName

[asterisk-users] basic installation

Hi I have just joined this group and i need to know what it take to have asterisk setup. What are the requirement. I ahve to submit a project for my college on VoIP and thouhg Astrerisk would be a great platform . Any help will really be great !!! -- Regards Agnello Dsouza

[asterisk-users] which phones to use ??

I am following the tutorial give at this link http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm but it does not mention the phones that i need to use could i use any USB phone !!! ??? thansk !! -- Regards Agnello Dsouza www.linux-vashi.blogspot.com www.bible-study-india.blogspot.com

[asterisk-users] asterisk queue agent problem

Dear I have queue setup arround 10 agent setup now what happend when call inter in queue and queue transfer call available extension but suppose extension A call to extension B ( ineternal sip call ) that time anycall come into queue and suppose queue transfer call on A

Re: [asterisk-users] Cisco 7965g and asterisk

On Fri, 2008-02-29 at 11:09 +, Nuno Fernandes wrote: Hi, We've just bought a new cisco 7965g and web are trying to connect it to asterisk. I've bought smartnet and downloaded [snip] How can i install the sip firmware? You need to setup a tftp server, put the 8 sip firmware files and

Re: [asterisk-users] basic installation

On Fri, Feb 29, 2008 at 12:33 PM, Agnello George [EMAIL PROTECTED] wrote: I have just joined this group and i need to know what it take to have asterisk setup. What are the requirement. I ahve to submit a project for my college on VoIP and thouhg Astrerisk would be a great platform . These two

Re: [asterisk-users] which phones to use ??

On Fri, Feb 29, 2008 at 1:12 PM, Agnello George [EMAIL PROTECTED] wrote: but it does not mention the phones that i need to use could i use any USB phone !!! ??? I would recommend you start by using free softphones like X-Lite, Gizmo project, Zoiper. Then, when you're ready, choose a

Re: [asterisk-users] Problems with removing zaptel

Hi, Will have a look. many thanks. On 2008-02-29 at 11:35 Paul Hales wrote: /etc/modprobe/blacklistor similar PaulH On Fri, 2008-02-29 at 00:30 +0100, Christian wrote: Hi all, Using the latest test version of Debian but when I have done modprobe -r and removed a few of the zaptel

Re: [asterisk-users] Problems with removing zaptel

Hi, Many thanks for that info, will give it a try. All the best, Christian On 2008-02-29 at 09:09 Tzafrir Cohen wrote: On Fri, Feb 29, 2008 at 12:30:49AM +0100, Christian wrote: Hi all, Using the latest test version of Debian but when I have done modprobe -r and removed a few of the zaptel

Re: [asterisk-users] TDM400P dialout problem

Anthony Messina wrote: i'm looking forward to 1.4.9.2, but am also concerned about http://bugs.digium.com/view.php?id=12099 as i saw this error with 1.4.9 and 1.4.9.1 on both platforms. The messages in bug 12099 are *not* errors, they are annoyances only. The latest SVN branch 1.4 code of

Re: [asterisk-users] asterisk queue agent problem

Hi Satish You would want to investigate Local channels on Asterisk for this. Garth Garth van Sittert BSc (Physics Computer Science) - Main: 08600 BITCO Phone: +27 (0)11 875 6900 Fax:+27 (0)11 875 6901 Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] MSN:[EMAIL

[asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init

Hi all, When I try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init. I've added #include curl/curl.h so the code compiles fine. this error is generated by the linker, even though func_curl.c is compiled and linked with no errors My asterisk machine have

Re: [asterisk-users] Pattern matching....

Tony Mountifield wrote: In article [EMAIL PROTECTED], Eric Wieling [EMAIL PROTECTED] wrote: No that will not work. You would want three exten = lines, one for each area code. And if you have a lot of common dialplan that you don't want to duplicate between the three extension patterns,

Re: [asterisk-users] Asterisk as useragent registered using 2 accounts

Thanx for the tip. It has erased the problem i was having using authentication but another problem has arised. i am now able to call with only one user from AST1 to AST2. If i dial using the other user, my AST2 shows the following warning and responds with a 403 forbidden sip response:

[asterisk-users] load balancing and high availability

I am evaluating the best way to make a high avail and load balanced system. I have two identical asterisk servers. Most clients are SIP phones. The only special hardware I have on both systems (they are identical) is: 1 E1 PRI card and 1 4-port BRI card. I have 8 ISDN lines so 4 go to each pbx

Re: [asterisk-users] when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init

On Friday 29 February 2008 08:10:40 Prashant Sharma wrote: When I try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init. I've added #include curl/curl.h so the code compiles fine. this error is generated by the linker, even though func_curl.c is

Re: [asterisk-users] TDM400P dialout problem

Anthony Messina wrote: with 1.4.7.1, i had no problems with either x86_64 or i386. with 1.4.8, i386 worked, but x86_64 did not. with 1.4.9 and 1.4.9.1, neither worked. i use the rpms from atrpms.net for fedora 7 i'm looking forward to 1.4.9.2, but am also concerned about

[asterisk-users] Gtalk with asterisk

Hi, I have been working with Asterisk for the ivr functionalities in the past. I am interested to implement the Jabber - Gtalk in asterisk. For which i installed the iksemel but this didnt help me out. I couldnt find the res_jabber.so file any where in the asterisk source directory. Still when

Re: [asterisk-users] Listening to Allison voicemail prompt on SIP phone causes [pop] sounds.

Ok so I'm not going crazy then. I filed a bug report. http://bugs.digium.com/view.php?id=12093 -Original Message- From: Trevor Peirce [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 27, 2008 6:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Re: [asterisk-users] Detecting SIT (Special Information Tone) on outbound calls

For reference of SIT please check http://en.wikipedia.org/wiki/Special_information_tone Regards, Sanjay. - Original Message - From: sanjay rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Sent: Friday, February 29, 2008 8:35:08 PM (GMT+0530) Asia/Calcutta

[asterisk-users] bugs.digium.com

Tracker seems to be down. -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue

Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but

Re: [asterisk-users] bugs.digium.com

On Fri, Feb 29, 2008 at 3:38 PM, Doug Lytle [EMAIL PROTECTED] wrote: Tracker seems to be down. Can't be! Mark once told me, The bug tracker is never on vacation! after I chided him on how much he worked when on vacation. ___ -- Bandwidth and

[asterisk-users] Detecting SIT (Special Information Tone) on outbound calls

Is there a way to detect SIT (Special Information Tone) when making an outbound call. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Listening to Allison voicemail prompt on SIP phone causes [pop] sounds.

shadowym wrote: Ok so I'm not going crazy then. The jury is still out on that issue! John Novack I filed a bug report. http://bugs.digium.com/view.php?id=12093 -Original Message- From: Trevor Peirce [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 27, 2008 6:46 PM To:

[asterisk-users] Request for testing: New wctdm24xxp and wcte12xp drivers.

Hi All, This is a request for testing for users of Asterisk 1.4 or 1.6 with any of the following Digium VoiceBus based cards: TDM2400P, AEX2400, AEX800, TDM800P, TDM410, TE120P, TE121, and/or TE122. From a practical standpoint, this branch should allow these boards to work in more systems /

Re: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue

Oh yes! This has been killing us for about a year. We've had several conference calls with my phone vendor and Polycom and it's still not fixed (or even determined why it is happening). Polycom keeps saying, upgrade to the next version of the firmware. We upgrade, still a problem. (again, for

Re: [asterisk-users] Gtalk with asterisk

Hi Naveen, For which i installed the iksemel but this didnt help me out. I couldnt find the res_jabber.so file any where in the asterisk source directory. Still when i run the command make menuselect the channel driver chan_gtalk shows xxx (dependencies not met). How can i register gtalk with

[asterisk-users] IAX2's JB and DTMF

We've moved within the last two months to Asterisk 1.4.x All remote facilities are connected via highspeed (9mbit) connections (Over OpenVPN) to a central Asterisk box, acting as a voice router, that funnels all calls into our Avaya Definity G3R via PRI. When corporate employees visit the

Re: [asterisk-users] SPA3102 registration problem

Quoting Mandeep Singh Bhabha [EMAIL PROTECTED]: what i did to configure SPA3102 is ... My problem is that normal SPA3102 configurations just don't seem to work. I can't even get the FXS port to register. I'm beginning to suspect that my unit is defective. Here's why: If I configure

[asterisk-users] Skewed RTP timestamps in SIP calls on Asterisk 1.4.18

Last week I migrated some of our servers to Asterisk 1.4.18 and we started seeing audio drops of several seconds during SIP calls. After investigating it we noticed that Asterisk was increasing the RTP timestamps abnormally during a conversation. I'm including a text file with a subset of the

Re: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN

As far as I can tell, with Polycom phones you cannot do what you're asking (which is for the PC and the phone to be in the same VLAN while the PC is connected to the phone). I don't know how they handle it when the voice frames are untagged, but they definitely won't pass tagged voice frames to

Re: [asterisk-users] which phones to use ??

For your own sanity's sake, steer as far away from Grandstream as possible. The firmware is appalling and isn't improving a great deal. They make great steps in one area while another gets worse and worse. randulo wrote: On Fri, Feb 29, 2008 at 1:12 PM, Agnello George [EMAIL PROTECTED]

Re: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN

Have you set the VLAN tag on the phone? CP Lee, John (Sydney) wrote: Hi all, I have been googling and testing without any luck and would appreciate any guidance from anyone. A port has already been configured on the CISCO switch with the following: interface FastEthernet2/0/1

[asterisk-users] callpark feature in ABE?

Hi All - Anyone know if the callpark feature is in ABE? Is there a comprehensive list of the differences between ABE and the open source version? I've only seen a bullet-point chart which has no real detail. Thanks, Noah ___ -- Bandwidth and

Re: [asterisk-users] which phones to use ??

On Sat, Mar 1, 2008 at 1:07 AM, Rob Hillis [EMAIL PROTECTED] wrote: For your own sanity's sake, steer as far away from Grandstream as possible. The firmware is appalling and isn't improving a great deal. They make great steps in one area while another gets worse and worse. I know a lot of

[asterisk-users] Help asterisk connectivity with MS SQL

hi all I want to connect my asterisk system with a MS sql .I have done it with MYSQL but i want to connect it with MS SQL.I have tried a lot but not getting anything.Can anybody help me on this.' Thanks in advance Rahul ___ -- Bandwidth and Colocation

Re: [asterisk-users] which phones to use ??

On Sat, 1 Mar 2008, randulo wrote: On Sat, Mar 1, 2008 at 1:07 AM, Rob Hillis [EMAIL PROTECTED] wrote: For your own sanity's sake, steer as far away from Grandstream as possible. The firmware is appalling and isn't improving a great deal. They make great steps in one area while another