similar to: No Audio with SIP to only one provider when switching servers

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

I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them. Here is what I have in my sip.conf: [stanaphone] type=friend secret=pAsSwOrD ; skewed for this message. username=3475341914 host=sip.stanaphone.com

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

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

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

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

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

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

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

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

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

Has anyone played with this phone? i cant seem to get it to work properly, i manged to get it registered and can make calls from it, but i havent been able to make it receive calls. Weird thing its that if you make a call from it and while you are on that call you dial its number does calls go thru in second line, but as soon as you terminate both calls it wont recieve any calls again. Heres

Ok, this is highly confusing. hestia*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 2944030 2944030 oneeighty_start No RFC3581 2944035 2944035 oneeighty_start No RFC3581 sip users (type=friend) are in sip.conf. I have nat=no

Steve, 1) go to /etc/asterisk 2) open modules.conf for editing using vi 3) add this line: noload=pbx_wilcalu.so 4) Save the file 5) Restart asterisk Lightup the candles, open the Cabernet Savignon ( or whatever your prefernce) and call your girlfriend. ;) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com

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 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

Hi all, Is there a possibility to set the codecs Asterisk will choose in the dialplan ("exten=>" statements or their contexts) instead of sip.conf? My problem is that I connect my SIP phone with several providers (Nikotel, Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these providers offer the same set of codecs. I'd like Asterisk to use the same codec for the

I was previous using Asterisk 1.2.9.1 and decided to get some real servers outside of my house. It was time for Asterisk 1.4.4. I figured since all the conf files were in /etc/asterisk form the old box, i'd just copy tha directory over to the new server. My SIP DID AGI stuff worked, except running 'asterisk -r' doesn't. It tells me ' Unable to connect to remote asterisk (does