similar to: Call from '6000' to extension rejected becauseextension not found

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Hi, Using a couple of sip phones and using asterisk to connect them to a single sipgate.de account. if I call a mobile I have no problem makeing conversions. If the mobile rejects the call (by pressing hangup while it rings), something strange happens: the following is seen in the logfile, everytime a rejected mobile call happens: ----------------- Mar 20 22:52:29 WARNING[4682]: Forbidden

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

Dear All, I am trying to recieve call from inbound proxy then route to internal peer (localhost) and then route to outgoing sip proxy but it failing with subject error. Can any one please correct me what i am doing wrong in below config. SIP.conf [Inbound] type=peer context=introuting host=184.107.XXX.XXX disallow=all allow=all [astinside] type=peer context=introutingB host=localhost

I went from asterisk 1.4.22 (which was working) to SVN and I am getting the message rejected because extension not found... How can I modify the print statement in chan_sip.c line 18388 to include not just the extension but the context its trying to find my extension in??? ast_log(LOG_NOTICE, "Call from '%s' to extension"

Hi All, how are you? I would like to know from you if the problem can be below is a BUG of the asterisk-1.4.21. I did an upgrade version of asterisk-1.2.18 for the version of asterisk-1.4.21 and now, when users try to sip friend outgoing calls through Polycom IP 330 appliances can not be the traditional way or with the telephone handset in his hand and digit dialing digit to receive the following

Hi all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic

Hello: I'm using: - samba-common-3.0.9-1 - kernel 2.6.5-1.358 - FC 2 - openldap-servers-2.1.29-1 We're running an NT4 domain using an LDAP backend, and everything was running fine until recently. The first thing that I noticed that new users were suddenly being assigned SambaSID's that were previously being assigned to machines. Previously: Typical User Entry: uid: john

Hi, I am running a couple of Samba / LDAP servers. While they all do work fine, I get a message like this on all of them when I run pdbedit -L -v: Unix username: administrator NT username: administrator Account Flags: [UX ] User SID: S-1-5-21-XXXXXXXXXX-XXXXXXXXXX-XXXXXXXXXX-21000 init_group_from_ldap: Entry found for group: 512 lookup_global_sam_rid:

Hi. We have an odd problem with incoming SIP calls. I have attached a SIP debug log, with some asterisk verbosity as well, demonstrating the problem, below. Is this a known bug? Vital stats: - Asterisk 1.2.3 - Sipura SPA-841, SPA-941 phones - Fedora core 3 The problem manifests itself with these symptoms: - an internal SIP extension receives a call from our PRI - the SIP phone answers the

Return-path: <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx> Envelope-to: xxxxx at xxxxxxxxx Delivery-date: xxx, xx xxx xxxx xx:xx:xx +xxxx Received: xxxx [xxx.x.x.x] (xxxx=xxxxxxxxx) xx xxxxxxxxx.xxxxxxxxxxxx.xx xxxx xxxxx (xxxx x.xx) (xxxxxxxx-xxxx <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx>) xx xxxxxx-xxxxxx-xx xxx xxxxx

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

I have a data set with observations on 549 cities spanning an 18 year period. However, some of cities did not report in one or more of the 18 years. I would like to implement the procedure suggested by Wooldridge section 17.1.3 in his "Econometric analysis of cross section and panel data" to correct for attrition. For example the table below indicates that the 3rd and the 7th cities in

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Thanks, Henry. It is very helpful for me. I also deleted the DIAL option "r" in our dial out trunk which fixed the problem. Dial command option r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Without this option, Asterisk will generate ring tones automatically where it is appropriate to do so; however, "r" will

I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail.

On a samba 4.5.1 domain controllers (PDC and BDC), classic domain , LDAP is used as the backend for both user accounts AND for the idmapping of trusted domains . Partial smb.conf below workgroup = THISDOMAIN security = user passdb backend = ldapsam:ldap://xxxxxxxxxxxxxxxxx idmap config * : backend = tdb idmap config * : range = 5000-6000 idmap config THISDOMAIN :

I have Samba 3.0.9 running on SuSE 9.2, 2.6.5-7.111-smp kernel. SAMBA is a PDC using OpenLDAP as a passdb backend.??Workstations?are?combination?w2k SP3, SP4, and Windows XP SP1. The problem I have is with profile synchronization. If?a?user?obtains?a?file that has a blank modified time, Windows substitutes Jan 13, 2038 for the

Hello, I have a SLES10 64bit machine running samba 3.5.12. i have configured a domain (TASC), and set the servers netbios name to TUX. Samba is using the tdbsam backend. Now I have add machines to the domain, and users can logon using their domain accounts TASC\user. However, if the network is disconnected, in the case of laptop users going offsite, they cannot logon anymore. I have now