similar to: Running SIP on multiple ports

Hi I had asked for some help with the Asterisk Assistants http://www.voip-info.org/tiki-index.php?page=Asterisk+Assistants+for+MacOSX and many have offered assistance with translations which I am grateful for and like to say thank you again. However, there hasn't been a single response from a graphic designer to offer help with a custom icon. Are there any graphic designers on this list at

I have seen a number of people in this newsgroup asking for information regarding asterisk interworking with Vocal. I was able to configure Vocal and Asterisk so that calls originating from vocal can land on an extension in Asterisk. I would like to share this info with the group The scenario that I tested was as follows. A call was originated from extn. 1001 on Vocal and the call was made to

I'm trying to get Asterisk to talk SIP to Vocal and so far have only managed to get it partially working. Calls in from Vocal are working fine but outbound calls aren't. In sip.conf I have: [ivv] secret=SECRET username=08452416761 host=sip.intervivo.net fromuser=08452416761 externip=mt104.dyndns.org nat=yes canreinvite=no reinvite=no notransfer=yes In extensions.conf I

Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to

A better explanation can be found here... http://www.digium.com/index.php?menu=faq#TDM%20&%20Analog_0 > -----Original Message----- > From: Benjamin on Asterisk Mailing Lists > [mailto:benjk.on.asterisk.ml@gmail.com] > Sent: Monday, September 27, 2004 11:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] FXO question

Does anybody out there have any evidence that Sipura support is still in operation? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.

Re-sent because had wrong subject line on first post, sorry. I have just installed * for a small office (P4 3Ghz, 1MB RAM, RH9). It's replacing an analog PBX, and for now, all incoming calls arrive on 10 FXO's. Outgoing calls are via Voicepulse. Phones are SIP, Cisco 7940G's. My problem is random disconnects on both incoming and outgoing calls. The phones are behind a firewall;

Ming-Wei Shih wrote (Re: [Asterisk-Users] Best Linux for Asterisk) > I am running * CVS head on Gentoo/i586 > and Gentoo/Sparc64 (US60 2x450/1GB RAM), > they are running great. > > On sparc64 * does not compile out-of-the-box, > some hackings in the Makefiles are needed. Great stuff. Please, can you share your adjustments to the Makefiles with the community?! If you don't

Logan O'Sullivan Bruns wrote: > I know Solaris isn't a well tested platform and I did have to make > some minor code changes to get to compile on my sun box. Well done! We need more momentum for Asterisk on non-Linux platforms. Building a community around Solaris much like there is a community around BSD, would be very helpful. This will only happen if Solaris users start sharing

>When I call a SIP user, the phone should ring in more than one >extentions. Also more than one phone should be able to register with >asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for

Hi Most of the threads in the list archive relating to X100P and hangups are about not detecting hangups. We have got the opposite problem. We have experienced an increased number of false hangups when connecting an X100P to an analog port of an ISDN terminal adapter. It happens more frequently on incoming calls than it does on outgoing calls. Often hangups occur after about 3-4 minutes into the

Anyone using the recently MAC OS X ? Version of asterisk ? Thanks, Francisco Perez-Landaeta > From: asterisk-users-request@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT) > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users Digest, Vol 1, Issue 5082 > > Send Asterisk-Users mailing list submissions to >

I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and line 2 on 5061. The SPA2000 is on IP address 192.168.17.6, and the asterisk box is on 102.168.17.2. Both SPA2000 ports(5060 and 5061) share the same IP address. Every minute I repeatedly get the following output: SIP Debugging Enabled 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.17.6 SIP/2.0 Via:

My apologies for the off-topic post ... No matter what settings I try, when I dial in to the SPA-3000 on the PSTN line, it picks up the call and immediately gives me a fast busy tone then hangs up. The info tab says under PSTN Line status: Last PSTN Disconnect Reason: PSTN Disconnect Tone which seems to indicate that the SPA thinks the caller has hung up. Since I am in Japan, it is possible

Hi anybody who would like to test drive a pre-release of the first OSX Assistant please visit the Wiki ... http://www.voip-info.org/tiki-index.php?page=Asterisk+MacOSX+Support if you find any bugs please let me know by email: benjamin (at) sunrise-tel (dot) com. Please take into account that this assistant is in the tradition of Apple's setup assistants which are always trading ease of use

I think I need to clarify what I meant by custom icon for the Asterisk Assistants in my earlier posting. On the Mac an Assistant is what the Windoze world calls a Wizard and there is a generic icon for it - the front of a dinner suit with bow tie, the one you can see on the Wiki. However, many of Apple's own assistants have a little mark in the lower right corner of the generic icon which

Hi for those who are unhappy with whatever mail reader arrangement they have reading the mailing list, I'd like to share my experience using Gmail, which I have been using for about a week or so now. I find Gmail to be excellent for the mailing list. It doesn't feel like a web mail application at all. The threading works perfectly. Responding to the list keeps the threads intact. It

A brief update on a few things which should be interesting to anybody who's got a Mac running OSX ... 1) JPT, integrated desktop dialer now with Asterisk support (and X-Lite too) Jon Nathan, the author of Jon's Phone Tool (JPT) has just released version 2.0.4 which supports dialing through local and remote Asterisk servers. It also supports dialing through a locally installed X-lite.

I was really looking forward to Asterisk 1.0 et al, but it is a major disappointment. I have never experienced any Asterisk release that was interacting with Digium hardware so unreliably. Asterisk hangs up on every outgoing PSTN call (via Zaptel) as soon as the call is being picked up at the other end. I have tried various X100P (original Digium) cards, various phone lines and just about every

Andrew Kohlsmith wrote: >I wasn't talking about bandwidth but rather lengthy >Dial() commands... > >exten => s,1,Dial(SIP/someuser&SIP/someuser&SIP ...... > >kind of thing... seems awfully unwieldy. That's why you would stick the members into a global variable [globals] DIYCALLGROUP => SIP/111&SIP/112&SIP113 etc. then dial using