similar to: SIP Registration --Giving up forever after very short network outage.

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

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 else seen this with recent CVS? Asterisk CVS-12/05/03-14:25:13 Dec 9 13:22:46 NOTICE[10251]: File chan_sip.c, Line 2981 (sip_reg_timeout): Registration for '^B@117.0.0.0' timed out, trying again Dec 9 13:22:46 WARNING[10251]: File chan_sip.c, Line 428 (__sip_xmit): sip_xmit of 0x81512ac (len 320) to 117.0.0.0 returned -1: Invalid argument Dec 9 13:33:07 NOTICE[10251]:

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Hello, I have a problem with an LAN-Server behind an NAT-router. Asterisk Version 1.2.1 or 1.2.2 doesnt matter 10 minutes after starting Asterisk I loose all registrations at external SIP-proxys. The reason seemed to be that Asterisk send every second an request to every sip-proxy "Request: OPTIONS sip:sip.domain.tld". Every request is responded by the sip-proxy. After some minutes

hello all, now, i want to do configuration to make sip client have extension on my asterisk.but i have a problem with registration of sip client. *CLI> May 31 13:58:01 WARNING[4927]: chan_sip.c:886 retrans_pkt: Maximum retries exceeded on call 4b5cbb235a46d6ee0bcd278c1e294105@192.168.8.125 for seqno 115 (Critical Request) May 31 13:58:15 NOTICE[4927]: chan_sip.c:4585 sip_reg_timeout: --

Hi all, I am using * as a PBX for a Broadvoice VoIP account. It had been working well since about last November, although not perfectly (similar disconnection problems, although I am pretty sure it had to do with my PPPoE setup, but I think these issues were resolved). As of a few weeks ago, though, I started having serious problems. Basically, I can start up * and connect to Broadvoice and

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

Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout: -- Registration for '1234561234@xxx.xxx.xxx.xxx ' timed out, trying again (Attempt #9) Is there something I can

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

The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and

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

(hardware in my computer: linux, asterisk, x100p, grandstream budge tone-100 ) Hi, When i run #asterisk ?v It show me a messages but when i try to incomming the call it show me that. Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'me@192.168.0.6' timed out, trying again Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration

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

Hi..I'm having trouble registering my asterisk box with FWD....It worked the other day. I also have an individual Grandstream phone which registers fine right now. I looked at the archives and saw the thing about the maximum retries limit to 5...but since my Grandstream phone seems to register on the first try, I'm thinking the problem lies elsewhere. Any ideas? sip show peers

Hi there, this is now the second time I've seen an issue like this with 1.2.7.1, the first time it was a DNS hickup, today its some Internet congestion: When one (!) or more register statements in sip.conf fail the entire Asterisk becomes very unresponsive and does not accept registrations from local phones. This is bad because I could live with being temporarily unreachable thru my

Hi people, My asterisk wont register with any sip providers, I have tried three different but they all end up with: Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout: Registration for 'whatever@provider.tld' timed out, trying again There is no firewall and my server has a public IP. Could this be a Asterisk problem? -Fredrik vK

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