search for: stanaphon

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=34753419...

...ry 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 incoming works ok so I don't think its any issues, and the machine is the DMZ of the adsl router so it should be forwarded for everything. These are the relevant snips of the file and the console output. ------si...

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

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 assured that their...

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 Linksys WRT54GS router (latest linksys firmware) acting as DHCP and NAT for my home equipment. I have tried forwarding the SIP and RTP ports to the asterisk machine, and I have also tried putt...

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

...from source on that machine (as I did on the old server). My firewall on the new machine is configured identically to the old one as well. All of my IAX connections just worked. All but one of my SIP connections just worked as well (which is why I can't believe it's a firewall issue). StanaPhone, which I use for 2 incoming DIDs, registers correctly, and rings my phones correctly when a call comes in. However, once answered, there is dead silence in both directions, on 100% of the calls. There isn't any problem on StanaPhone's side (which has provided a _fantastic_ service eve...

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....

...also have a SIPPhone number, which (obviously), connects via > SIP. I called that number from the outside, using one of their > "Access Numbers", and my phone rang and I heard audio in both > directions (this with the firewall back on), so SIP definitely > works, just not with StanaPhone. > > Then I connected from another server that I run, which is behind a > NAT router. That server is running 1.2.18 (as is the one that > isn't working, but is on a public IP). Audio works perfectly with > this one. > > To my knowledge the only difference between them is t...

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

...-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 accounts as well - all work flawlessly) I have a simple dialplan - an incoming call rings all phones and goes to voicemail if not answered. When I dial '8' followed by a number - the call routes out via Stanaphone fine. No issues. When...

...; This section is because i'm behind nat externip = x.x.x.x ;Outside address localnet = 10.73.73.133 ;Inside address localmask = 255.255.255.0 ;Inside subnet context = sip ; Default context for incoming calls register => ##########:secret@sip.stanaphone.com/1000 register => ##########:secret@sip.provider.net/4078 register => ##########:secret@sip.provider.net/4077 [stanaphone-out] ;works!!! host=sip.stanaphone.com context=sip type=friend dtmfmode=rfc2833 canredirect=no disallow=all allow=ulaw insecure=very username=se...

...ternational phone number conventions. I have my contacts in Outlook, with the numbers represented as: +<countrycode> (<area code>) <numberpart> <numberpart> eg: +44 (20) 7834 1234 or: +1 (801) 555 1234 I'm using the SJphone softphone, doing my testing through the Stanaphone SIP/PSTN gateway, and have Asterisk in the middle working fine for everything else. SJphone can pick up the numbers from Outlook, and send them to Asterisk, but Asterisk is finding that Stanaphone is barfing on any number that includes a space or a ( or ). If I manually enter the number without...

...ial internal SIP extension numbers. This of course leaves me with no SIP [brackted] section of which to use for outbound calls of which I'd love to eventually get working. Am I doing this right at all???? or am I headed completely in the wrong direction here? I also tried this with a FREE Stanaphone account and get a very similar but strange result.... IN both cases adding this section to sip.conf result in my calls terminating at the SIP provider voicemail system instead of coming into my asterisk box here. A side note: Not that it really matters but here's what I get from my prov...

...der 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 stanaphone, but somewhat more reliable) in *any* rate-center? TIA

...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 providers (stanaphone, simpletelecom etc.) and the quality is OK. tcpdumping connection with both providers showed that G.711 alaw/ulaw was in use. Some Grandstream phones are also hooked into that same Asterisk pbx and using G.711. Those devices voice quality is superb. Firmware installed in Zyxel Prestige 2002 is...

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 -------------- An HTML attachment was s...

...HEAD (about two weeks old) Aug 24 19:08:13 NOTICE[7124]: chan_sip.c:4701 sip_reg_timeout: -- Registration for '734nnnnnnn@sip.main-provider.net' timed out, trying again (Attempt #8) Aug 24 19:08:16 NOTICE[7124]: chan_sip.c:4701 sip_reg_timeout: -- Registration for '516nnnnnnnn@sip.stanaphone.com' timed out, trying again (Attempt #11) Aug 24 19:08:16 NOTICE[7124]: chan_sip.c:4719 sip_reg_timeout: -- Giving up forever trying to register '516nnnnnnn@sip.stanaphone.com' Thanks for your help !!! Steve

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 provider side as well as of the device side, to prevent codec translation. Unfortunately, Asterisk seems to negotiate the codec for the device and for the p...