similar to: User registration failure bug ?

Hi list, Currently, one of my phones registers fine, and the other does not, though for me they have the same config... Can somebody help debug/understand why? The logs in asterisk say: [Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 <sip:IMSI20830061xxxx at 127.0.0.1>' failed for '127.0.0.1' - No matching

do not know why. Anybody has a clue what could be wrong ? Is this a bug ? [I rebooted asterisk, and now it works.] Regards Axelle. Logs of failed registration: > sip show users Username Secret Accountcode Def.Context ACL NAT IMSI208011234567890 sip-local No RFC3581 IMSI208302141472352 sip-external No

Hi list, I have searched through asterisk command lines but haven't found how to do this: - can I list the phones (callerid or IMSIs?) currently registered ? If I do "dialplan show" that lists the configuration I loaded, e.g [ Context 'sip-local' created by 'pbx_config' ] '2102' => 1. Macro(dialSIP|IMSI1) [pbx_config] '2103'

I tried this [dongle0] ;audio=/dev/ttyUSB1 ; tty port for audio connection; no default value ;data=/dev/ttyUSB2 ; tty port for AT commands; no default value ; or you can omit both audio and data together and use imei=123456789012345 and/or imsi=123456789012345 ; imei and imsi must contain exactly 15 digits ! ; imei/imsi discovery is available on Linux only

Dear List, I frequently use the data() function to load csv files (with separator ";") into R session, typically data(myfile) loads myfile.csv from my working/data directory into R. Now, in 1.4.0 version, everything works as expected, but with one difference: The values readed in older versions in "num" mode are now readed as "int" mode, converting the values

Hi All, I am using Asterisk for one of my projects in OpenBTS. I am having the age old problem of "extension not found" when try to make a call from one registered SIP phone to other registered SIP phone (two mobile phones connected to Asterisk via OpenBTS). The exact error thrown on Asterisk CLI is *"chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to

Dear R users, I'd like to make this data rem.y = c(-1,0,2,4,5) from y = c(-1,-1,0,2,2,2,2,4,4,5,5,5,5,5). That is, I need to remove repeated values. Here is my code, but I don't think it is efficient. How could I improve this? #------------------------------------------------------------------------ y = c(-1,-1,0,2,2,2,2,4,4,5,5,5,5,5) n=length(y) for (i in 1:n) #

Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: *Registration error: 404 Not found* Here my configuration file of asterisk: *[root at dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* *[root at

Apparently, the realtime system in asterisk is faulty. Implementing realtime, begins a host of seeding messages along with registration messages visible at the CLI prompt. This is not the case with .conf file configuration Unfortunately, it is not clear where the bug originates but is shows it's head while calling the register_verify function, (which there are 2 one in chan_sip.c and one in

Let's say I have two Asterisk boxes, A and B. I am trying to get A to do SIP registration on B, so an extension for A can dial SIP phones covered by B. If I examine the logs on B, if the registration succeeds, I am seeing a notice to that effect on B. But if the registration *fails*, i'm not seeing any message logged on B. Maybe I'm just not looking in the right place. Is there a

hi, i use xapian to index a txt file, it's size is 268M. i take each line as a document, and each line has two field like 13445511 | 111115151. the recored size is 10000000. the XAPIAN_FLUSH_THRESHOLD set 1000000. it takes 1026544ms to index the file, it is more slower than lucene. The lucene speed is about 40000 records per second. code: try { Xapian::WritableDatabase

Hi Folks, I am newbie to asterisk. Recentely I have installed asterisk on Linux Fedora 2 box. After reading some document, I tried to configure the server. When I connect to our server, SIP user-agent shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below. *CLI> sip show peers Name/username Host

Hi folks, I wonder if any of you out there are using Siemens S685IP base station(s) (with S68H handsets) on Asterisk and experiencing problems with SIP registrations where the SIP extensions do not ring and peers become unreachable after a period of time. Symptoms are rather sporadic, but as described, SIP extensions being unreachable from Asterisk perspective. Also experienced 'not

Hello list, I am new to this list and have been using Asterisk as part of my research project for about 2 weeks now. I would like to get your thoughts on a scenario that I am attempting at the moment. I haven't had luck until now. In this scenario, I am trying to register a SIP user configured on the zoiper client installed on a laptop, which is on the same Local Area Network, with the

Hi all Looking for a solution to create a flexible voicemail solution in Asterisk without the need to preregister the voicemail users (via databases etc etc). Scenario: All incoming calls are voicemail calls however the dialled number (called party) does not necessarily have a voicemailbox configured in the Asterisk system. I am looking for * to do the following: * Call comes in *

Hi list, I'm in central-europe and signed yesterday a broadvoice account. My Asterisk box is CVS 2005-08-25. Problem I face is: "Failed to authenticate on REGISTER to 'phonenumber@sip.broadvoice.com' (Tries 2)" then "Registration for 'phonenumber@sip.broadvoice.com' timed out" and finaly "Giving up forever to register

If anyone is experiencing this type of registration error: Registration from '<sip:40852@192.168.70.2>' failed for '192.168.70.25' try adding the following line to your modules.conf file noload => app_adsiprog.so This error is clearly asterisk trying to register with the phone and not the other way around. the app_adsiprog.so is only used to download adsi scripts to a

Thanks for the feedback ! I know the IP-address of my Asterisk-server. The WAN-interface of my Asterisk-box is set manually (ifcfg-eth1). I have port 4569 forwarded on my NAT/firewall. Strangely I have the same 'notice' when being attached directly to the internet (so no firewall in between). And set my WAN-interface to the IP I get from my ISP or even when obtained by DHCP. Doesn't

I have a very lite layout and attempting to get the SIP configuration set up initially before proceeding into other areas. VMware is running my Asterisk 11 on Ubuntu 12. Shouldnt I be able to at least ping the SIP provider IP? I run command "sip show registry" and do not see it set up. I run sip show peers and I do see an entry. I have not configured anything other then entries in the

Hi Sorry if this comes twice; i sendt first version from non-member address. I'm learning use Asterisk but cannot solve following problem: i have Asterisk v1.0.7 (DEbian) and Linphonec v1.2 (Debian). Every time i try to register within LAN i got 'Forbidden' message from Linphonec. Where to start searching for reason for this failure, is there more debuggin options