similar to: Stanaphone Configuration

[root@iw-asterisk zaptel-1.2.1]# make gcc -I/lib/modules/2.4.21-4.ELsmp/build/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/lib/modules/2.4.21-4.ELsmp/build/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/lib/modules/2.4.21-4.ELsmp/build/drivers/net/wan -I/lib/modules/2.4.21-4.ELsmp/build/include/net -DMODVERSIONS -include

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

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

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

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 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

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

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'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

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

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

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

I'm having problems in dialing numbers over SIP that include characters from the UK international 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

After two weeks of reading about asterisk and joining this mailing list, I finally decided jumping on the asterisk bandwagon... Asterisk rocks!!! I have a www.Stanaphone.com SIP (free) for incoming line and a www.VOIPJET.com IAX line for outbound. I also have a www.Vonage.com line (gives me 500 outbound minutes) and a Cingular cell phone (gives me 800 minutes) and I also use Skype fairly

Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to

Hi! I made the compilation of Samba with support for clustering as shown in the Wiki and ctdb.samb.org, however when starting the service with / etc / init.d / start services ctdb the error is returned. "You must configure the location of the CTDB_RECOVERY_NODE. But I've created a directory called / cluster_storage / shared / is appended in the smb.conf file path on your [homes] path = /

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 looking for some help in how to keep asterisk from doing this. If we loose Internet or routing to our upstream provider even for only a few short minutes asterisk quickly gives up & never tries again. I have to do a manual reload to get it to register with my sip provider(s) again before incoming calls are accepted. This is really bad as it causes us to loose the ability to get