similar to: SIP Registry Problems

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both set to register with the same account right now. I shut Asterisk down on

Dear friends, Asterisk's voicemail functions work fine for me, but I am having difficulty storing the voice messages inside mysql. My real-time CDR recording works so I assume the odbc connection is fine. The voicemail.conf I have is : [general] format = wav attach = yes dbuser=root dbpass=sqlpass dbhost=localhost dbname=asterisk odbcstorage=asterisk odbctable=voicemessages Asterisk shows

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them

Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen

hello, i try to compile * in uclibc enviroment (uclibc 0.9.19 ), but still getting following error does anyone know how to solve it ? regards Marian --------- gcc -g -o asterisk -rdynamic io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

Ever since upgrading to 1.6 I get messages like these. I want everything else that shows up, but is there a way to make all the dns messages go away? Ira > doing dnsmgr_lookup for 'gw5.telasip.com' > doing dnsmgr_lookup for 'sipconnect.ipcomms.net' > doing dnsmgr_lookup for 'proxy.ideasip.com' > ast_get_srv: SRV lookup for

Hi, I am simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the

It gets hard to read my logs when every time someone makes a phone call it displays long pages of "Dropping voice frame". Anyone encounter this before? Asterisk is bridging two SIP lines together, so the technology should be the same. Maybe I'll try allowing only ULAW. ************************************** Asterisk Standard debug (level 3)

Ive recently installed three servers with RHEL5u5. After some messing on the original, I got samba working with ADS authentication. I then went and got it working so that users could log in using their domain name & password to the box. I got this working with both no restriction, and ADS group restriction. I have left it on no restriction wheil I get these systems up and running. I then

Try as I might, I can not get incoming calls from ViaTalk to match against my user entry. I have both peer and user entries, and incoming and outgoing calls work, but incoming calls do not move to my in-viatalk context (they stay in the default context.) Has anyone else managed to get this to work? My user entry looks like: [viatalk-in] username=1407965XXXX context=viatalk-in type=user

>On 18/03/15 17:56, Jesper Koivum?ki wrote: >>/ Hi, />>/ />>/ I'm running a samba 4.2 server on RedHat5 and for some reason I can't />>/ seem to logon using the AD Users and Computers -tool. />>/ />>/ Whenever I try to connect to the PDC I get the following error: />>/ />>/ "The following Domain Controller could not be contacted:

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

Let's say I have a third-party device acting as a sip<-->pstn gateway, a cluster of three asterisk servers, and a teensy bit of dns knowledge. Let's now say those asterisk servers are a1.company.com at 192.168.0.1, a2.company.com at 192.168.0.2, and a3.company.com at 192.168.0.3. 1. If I setup round-robin dns like so: asterisk.company.com. IN A 192.168.0.1 asterisk.company.com. IN

On Tue, Nov 7, 2023 at 11:20 AM Marek Greško <marek.gresko at protonmail.com> wrote: > Hello, > > well I do not ask those who only guess, but those who know what is > asterisk expected to do when internet connectivity goes down. I did not had > a chance to make internet not to work yet, since it is needed. But > inspecting dns logs I found out that there started to be

Hello, well I do not ask those who only guess, but those who know what is asterisk expected to do when internet connectivity goes down. I did not had a chance to make internet not to work yet, since it is needed. But inspecting dns logs I found out that there started to be resolving for _sip._tcp and _sip._udp records for the provider's server. So apparently making hosts record make asterisk

Hello, I cannot call my server over the internet with SIP anymore. Even when I do a maximum logging on my firewall, I don't see packets coming from outside. I've tried it from an ekiga.net account and an sip2sip.info account. What could be wrong? I would expect incoming traffic on port 5060 UDP... The account is "paul at vandervlis.nl". This should connect trought DNS to the

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at