similar to: RE: Asterisk-Users Digest, Vol 18, Issue 131

Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect

That was exactly it! Thanks you VERY much! Mike ---- For the sip setting in sip.conf that setsup your voip provider add: canreinvite=no On 2/6/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote: > > Hi, > > I've had a bit of a problem with one way audio, and it happens exactly when > I believe it shouldn't (and works perfectly when I would guess I could

So I go into a new Apple store on Sat to buy some stuff for my Mini, and I notice some Snom 360's on the sales counter. Venturing a question, I ask, are they using Asterisk? Guys says yes. Cool! I said: What kind of box are you using. He points to a Mini sitting on the counter! 2 X cool! He's using a SIP-FX0 converter. Plug: http://www.mymacdealer.com great store in Alberta. Anyone else

So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; register=9738281625@sip.broadvoice.com:PASSWORD-GOES-HERE:9738281625@sip.broadvoice.com/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very

One of my buddies signed up with GV yesterday with a view to using them on his * server. Problem is that the settings they gave him don't work with asterisk. They do however work with X-Lite. Any ideas? He's using the settings outlined on my web page. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/

I've been banging my head with this all day. I today switched from a very old CVS build to AAH1.3 and so far everything has been easy. However I cannot accept calls from a previously working IAX trunk. I've set up an trunk with all the same credentials as before and can call the folks at the other pbx. However whenever they call me I tell them that I don't have an

I was at VON in Boston today and saw on the Digium stand a Cisco 7960 with a picture of Tux and the Asterisk log on its display. I WANT IT!!!!! Anyone know where I can download this file please? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hi Folks, OK, I'm running CVS Head as of about 3 weeks ago. I want to roll back to V1.09. Other than downloading the code, how do I do it? I thought someone once said that I have to delete all my modules or something? Thanks Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Does anyone have a collection of stupid hold music? Y'know, the sort of thing that would drive a person mad? Silly songs, repetative tunes etc? The best I can come up with is; I know a song that drives everyone up the wall I know a song that drives everyone up the wall I know a song that drives everyone up the wall And this is how it goes I know a song that drives everyone up the wall I

I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server

Hi Folks, Prior to upgrading my Zaptel sources everything was working fine. I have a X100P connected to my analogue line. The handset port of the X100P is connected to my desk phone's line 2 input. When the analogue line rings I see the CID on my line 2 but not from Asterisk on line 1 via the Cicso ATA. This used to work fine until I upgraded the sources. I get this when watching the

Hi Yawl, After about 6 months of prattting about I've convinced my boss that we should be installing * into our currently under constuction Data Center in Somerset NJ. There will be 10 permanent people and DR space for another 50. My plan is as follows; ATAComm dual XEON server with quad T1 board. A handfull of ATA's for fax machines, job lot of X-Pro softphones for the DR bit, Polycom

That's great.....it's a virus I tell you * is everywhere :) Viva la asterisk. Cheers, Dean > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Mark Phillips > Sent: Friday, 3 June 2005 6:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject:

Hi Folks, I have an IAX trunk link to a collegues house. I'm using AAH and he's got the latest CVS as of last Tuesday. Problem we're having is this; when I dial his extension 7201 (Pulver WiSIP phone) his * box sends me 1 ring and then Alison's busy message. If I call his 7202 extension (X-Ten Pro on a Win2K laptop) I get through but with only 1 way audio (me to him). Until

I downloaded the latest CVS a few days ago. It all compiled nicely on my new AAH platform. However, it won't start up. Investigation of my log files produces this; Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 WARNING[31473] loader.c: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol:

I just built it and now can no longer get incoming or outgoing service. It was working with CVS Head prior to my "downgrade". Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hi all, Anyone got any VoIP traffic shaping rules for m0n0wall that they could let me look at please? Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

I think I've asked this before and think that Matt had said something about this. Is there an LCDproc client for Asterisk available and if so how can I get a copy please. Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hi Folks, I've been playing with call recording for our support department which was kinda going ok until I spotted something odd in the CDR. None of the support calls are being entered into the CDR properly. I'm using mysql as the back end and Areski's web based front end and all was going fine. The problem seems to be that the CDR doesn't get populated with the destination