similar to: voicemail notify problem on sip extension

I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call

Enhanced OS. General rules I use: -Do not use SIP transformations (the VOIP tab), these cause random RTP = issues, and once you start forwarding calls between users, all things go = to heck. You are better off using NAT/qualify in your sip.conf. -Do not use SonicOS Standard (all new Sonicwalls should come with = Enhanced now anyway) as there is no method to increase the timeout for = UDP rules,

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so forth in 1.4.21.2) as *literal* backslashes to Mysql, so that Mysql gives a syntax error

I try to install astcc. Make install gives me: DBI connect('database=astcc;host=localhost','astcc',...) failed: Unknown database 'astcc' at ./astcc-admin.cgi line 67 DBI connect('database=;host=','',...) failed: Access denied for user: 'root@localhost' (Using password: NO) at ./astcc-admin.cgi line 60 Ignoring that (!!! ;-) ) and going to the

Is there some magic to compiling asterisk-addons-1.6.0 under Debian 2.6.18 and mysql 5.0? I am unable to get configure to recognize the existance of mysqlclient. Imparticular, when it gets to: checking for mysql_init in -lmysqlclient... it returns "no". For the past several releases, I've had to hack or otherwise coerce this to work, but this time, no amount of fiddling with

I have a Uniden wireless phone connected into Linksys/Supura 2100. It works well, except I never see any caller ID information displayed on the phone. Is that a setting in the 2100 that I'm missing, or is it an Asterisk setting or isn't it possible? Thanks, Jim.

vpbx*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 621/621 192.168.250.76 D N 5060 OK (65 ms) 626/626 192.168.250.109 D N 5060 OK (180 ms) 616/Ronald Softphone (Unspecified) D N 0 UNKNOWN 615/Ronald office 192.168.250.103 D N 5060 OK (41

Count me in too. -----Original Message----- From: sip [mailto:sip@intology.com] Sent: Friday, October 17, 2003 1:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in ----- Original Message ----- From: "Paulo Mannheimer" <paulohm@instant.com.br> To: <asterisk-users@lists.digium.com>

The message "Returned mail: Data format error" from Bounced mail, sent on 3/9/2006 10:47 was quarantined because it contained either an executable file, a batch file or a screen saver file. All of these types of attachments are considered security risks. Please consult your mail administrator who can release the message. This message was checked by MailScan for WorkgroupMail.

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No one else would step up to the plate so I did. Here is your list. HAVE SOME FUN! need any more just ask. Troy Settle wrote: > With all the discussion about licensing issues and the sort, I think it's > time for a full blown 3rd party application to work with Asterisk while at > the same time not causing Asterisk to become encumbered. For such a > project, I'm license

the link is at www.pawbell.com ----- Original Message ----- From: sip To: asterisk-users@lists.digium.com Sent: Thursday, October 02, 2003 8:57 AM Subject: Call it Asterisk-Addons and let us go have some fun? No one else would step up to the plate so I did. Here is your list. HAVE SOME FUN! need any more just ask. Troy Settle wrote: > With all the discussion about licensing issues

Everyone seems to be working on their own servers that are in there homes, offices and elsewhere. The only common thread is this list-serv. I have several * servers set up here in Austin, Texas. I propose we set up one of them with with all the list-serv members. At this time the calls would be limited to on-net calls only to other members. As the project grows and other members add servers we

I try to change something in ASTCC, but I am now totally blind, .... I hang on one line now. I changed: vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi 22c22 < # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN}) --- > # exten => _00XXXXXXXXX,1,DeadAGI(astcc.agi,${CALLERIDNUM},${TARIFF},${EXTEN}) 35c35 < # exten =>

Good day everyone, A Solution now exists for those of you still looking for the first model of Quad T1/E1 Interface Cards - the T400P and E400P. The T400P and E400P were manufactured by Digium until a few months ago, and were designed and released under the GNU Public License as the Tormenta II by Zapata Telephony. Varion, Inc. is now in full production manufacturing these boards. With the

Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab -> Dial Plans -> Dial Plan 8 (<S0:66610>) Should I put extension [66610] in sip.conf with a context in extensions.conf that

I have an * system running version 1.0.8 and it works mostly fine. I am using it as a virtual PBX and we share the box among companies. I have the calls all staying separate, we well as the companies' extensions, voicemail, etc. The only problem I'm having is with two accounts that use the same SIP termination provider. * seems to be confusing the sip contexts for the incoming calls.

Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis.

I tried to install astguiclient and it gives me for each follow page: Object not found! Looking into the apache log file I find: [Sun Feb 27 16:18:30 2005] [error] [client 192.168.250.108] File does not exist: /srv/www/htdocs/astguiclient/method=POST, referer: http://vpbx.elmit.com/astguiclient/phone_stats.php?extension=gs102&server_ip=192.168.250.20 I have uncommented the ### If you have

I have now a very strange situation. Asterisk dies exactly every hour at hour:09 !!! crontab has the entry: 10 3 * * * root /usr/sbin/asterisk-restart >/dev/null 2>&1 It seems that there is a time difference between mail server and asterisk server so that it might be "synchronized" to the crontab entry. *CLI> show version Asterisk CVS-HEAD built by root@vpbx on a