similar to: StanaPhone and Asterisks

Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip

I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble.... thanks Todd

I am having problems getting the latest CVS right now. A cvs checkout asterisk -t gets to this part and sits forever: S-> server_register(fpm-world-mix.mp3, 1.1, , , , , ) S-> Register(fpm-world-mix.mp3, 1.1, , , ) Anyone know how I can just skip the file? Travis Conway EFS, Inc. Information Technology Desk:?? (334) 215-6551 Mobile: (334) 391-4450 mailto:travis@homeoffice.quikpawn.com

Anyone having problems with Stanaphone registration today? I'm getting the following.. Nov 25 11:35:58 NOTICE[229390]: chan_sip.c:4053 sip_reg_timeout: Registration for 'xxxxxxxxxx@216.128.82.18' timed out, trying again -- Got SIP response 500 "Internal Server Error" back from 216.128.82.18

For some reason I now seem unable to control which codec is chosen. The problem happens with outgoing calls to Stanaphone. Even if I chose disallow=all and allow=ulaw as the only codecs it connects with GSM. Has anyone else got problems with these settings? Any suggestions? As I recalled it, such a setup would not establish a call if the ulaw-codec was not offered by the provider. Stanaphone has

They say on their website that they allow you to use your own device provided you give them the MAC address. Has anyone tried using * with it? Looks like they have quite a few rate centers and also phone support... Their website is horrible though... Just wondering, it'd be good to get user experiences from different providers other than IconnectHere and BroadVoice... -Chris

Hi, I am pretty new to all of this but was able to set up an asterisk server and have been able to succesfully connect to asterisk with x-lite as sip client. I have also connected asterisk to FWD (using iax2) and to voipjet (also using iax2). Now I am trying to connect asterisk to Stanaphone. It has to register as a SIP client but I am not being succesful at all. My asterisk server sits behind a

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

We are having lots of problems with stanaphone. It used to work ok, but now it's terrible. As of this moment, can't make outbound or inbound calls. Anyone has it working? Please provide sip.conf example commands.. Thank you -- Leandro Rzezak leandror@gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, After much searching of voip-info.org and google, I'm finally giving in and asking the list. The setup I have is this:- Single X100P card in a Debian system Inbound/Outbound POTS line connects to the X100P Sipura 2000 and Budgetone 100 on the LAN 1 Cordless and one conventional phone connected to the sipura Account on Stanaphone.com for eitherbound SIP calls. (I have other SIP

Hello, did you got your issue solved? I am suffering of the same issue.... On 4/28/07, Hadar Pedhazur <hadar@unorthodox.com> wrote: > > I snipped all of the previous data, as I'm trying to "boil down" > this problem to its essence... > > I turned off the firewall for a few seconds, and still got no > audio. For those that will be suspicious, the commands

Now that I have most everything actually working I've noticed that about every 3-4 days on average..... and at worse... Once a day my asterisk box seems to lose it's registered state with our sip provider and no longer will take any incoming calls. The caller simply hears a fast busy (reorder) If I do a reload at the command prompt all is well for another few days..... What I'm

I'm going to try and ask this again and keep it short and as too the point as I can while still providing enough info to be of use. PLEASE advise if I am going about this wrong or asking too much. I'm seriously doing my BEST to throughly read the docs and try a bunch of things BEFORE coming here to ask and possibly annoy. If is documentation that explains thsi process in terms that

Should I be concerned about this? It seems to only happen when my MoH switches songs. The songs sound as good as an 8k/s song would. Travis Conway EFS, Inc. Information Technology Desk: (334) 215-6551 Mobile: (334) 391-4450 mailto:travis@homeoffice.quikpawn.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does IPKALL still exist? I am after a free SIP trunk - who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -------------- next part --------------

Here's a question I haven't seen asked nor answered on this list: Is there a provider who offers incoming 900# services? I want to establish a 900# to be used in (about 60-70) domain registrations, to deter telemarketers from calling yet still comply with ICANN requirements for a valid phone number. Alternatively, does anyone know of a source for super-low cost DIDs (like free

Hi, Does anyone have suggestions hooking Zyxel Prestige 2002 to Asterisk? I have tested Zyxel Prestige with both supported codecs. Call with G.711 sounds very choppy and cracking. Almost can't understand a word. Today I installed G.729 support into Asterisk but unbearable voice quality remains. It's a little bit better though. I have tested that Zyxel ATA with some commercial SIP

I had a working 2-way SIP connection running until about 2 days ago, now my outbound calls are promptly blocked with a "403 Forbidden" error. Inbound still functions OK. Perhaps they are fingerprinting and blocking Asterisk access (I hope not). They do not answer their support mail, or questions on their own forum. I'm sure there are other Asteriskers out there who have

Could you please call it and confirm with me it's not working for you either? I should probably transfer my DID number anyways, if I could only get them to respond! Does anyone have a suggestion as to where to go in this situation? Possibly a place with high capacity concurrent incoming calls... -- Anything else, let me know. - Dominic "It is not the force of a stroke that makes fine

I have a lot of experience, all of it pretty good, with various Sipura products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into Asterisk as clients. Good sound quality, great reliability. I've tried two of the units named in the subject line, and frankly I'm frustrated. Calls usually start out OK, but within a brief period the sound goes totally to