similar to: Script to automatically update externip. Useful for a host with dynamic public IP

I have been reading about testing in RoR for what seems like hours now in search of the answer to what is probably a simple problem. Where is the best place to test the second action explained below? I have a controller (NetworkingController) with a method (create_network_segment) that makes use of two models (NetworkSegment and NetworkIpaddress). This particular method does two actions: 1.

I am running Trixbox 2.4 which has Asterisk 1.4.18-1 I have kind of followed: http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 I added to sip_general_custom.conf ;NEEDED!!! t38pt_udptl = yes I did not add this to the actual SIP extension, as I assumed this being general it applies to all sip extensions, and doing a sip show peer ext# did indeed come up with t38pt_udptl = yes

Hello guys. When I am trying to send fax through T38 to linksys SPA (properly configured etc. - I have tried it with other systems), I'm getting error and fax is not delivered. I'm getting this errors in asterisk.log: WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c: UDPTL creation failed WARNING[687] udptl.c: No UDPTL ports remaining then, couple lines down:

I am working on a SLES 10 cluster with R available on it. To better use my resources, I want to use Rmpi, but I am having a difficult time installing it. I have set the $R_LIBS variable correctly. The MPI libraries are available through the PGI compiler, but this R instance was compiled with GCC. The Rmpi package can be compiled and installed with %R CMD INSTALL Rmpi_0.5-5.tar.gz

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

Curious. I have a gentoo server running 3.0.23d that simply serves out shares. It is a domain member, but not a pdc. From another linux server, I can mount up shares without a hitch. But from a windows box, I keep getting prompted for credentials. I am not seeing anything substantial in the logs. SMB.CONF -------- [global] workgroup = UNICITY realm = MYREALM.MYDOMAIN.COM

One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they

Does anyone know why? ThePBX*CLI> -- Executing [310-456-7890 at from-trunk:1] Set("SIP/202.101.202.101-b763ce60", "__FROM_DID=310-456-7890") in new stack -- Executing [310-456-7890 at from-trunk:2] ExecIf("SIP/202.101.202.101-b763ce60", "1 |Set|CALLERID(name)=310-456-0987") in new stack -- Executing [310-456-7890 at from-trunk:3]

I think I must be doing something wrong... when a form returns params to the controller, empty fields appear to be returned as blank, not nil. so what''s the point of an :allow_nil=>true setting on a validation? it fails because blank is not the same as nil. i would expect either returned empty fields to be nil, or :allow_nil to really be :allow_blank. otherwise i have to do a search

OK, I'm certain that this topic has been beat to death, but I need some assistance. I am trying to migrate to a SSO for the majority of our workstations and servers within our organization. I am currently trying to integrate a Gentoo Linux workstation to authenticate to the AD server. Once I get the process nailed down, I'll be moving on to bigger and better things... Prior to

Hello, I have an issue that I can't sort out. Issue: Just applied the latest round of patches that brought me up to this Samba version and suddenly end-users are being prompted for authentication when attempting to access shares on this CentOS box from their Windows Vista, 7x86, and 7x64 workstations. Problem: I am new to Samba and seem to not be connecting the dots Layer 1: I can ping

Hello, What is the set of configuration that should be done in the Asterisk 1.0.8 using FreePBX that can allow a simple video call between two extensions? Thanks in advance. [http://www.ericsson.com/shared/images/Email_line.gif] JULIO ARAUJO TE ENGINEER MS Ericsson ITTE & Test Environment S?o Jose dos Campos, Brazil Phone +551239084121 SMS/MMS +551281150089 julio.araujo at

Hi All, Is it possible to specify more than 1 localnet? I know this is an odd question. I have a customer that has multiple sites linked by VPN. Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24 We want to allow some access to the public IP address at the main site. For this to work I need to use the externip and localnet directive. If I do this it rewrites the SDP with the

Hi, Is there some parameter that I should pay attention to when using externip parameter on sip.conf? I ask this because after using sipsak I've noticed maybe the reason that i don't get voice in one direction could be because at the SDP/SIP messages are references to the 192.168.1.50 (server internal ip) and not the ip from the provider. Is there a way to alter SDP messages to correct

Can i use a domain name instead of an IP address for externip (sip.conf) Because im using dynamic dns. Not sure what i'm trying to achieve as yet but, i want to know if it is possible?

Hi, What is the equal option for externip in asterisk 13 with pjsip. I have tried external_media_address=XX.XX.XX.XX external_signaling_address=XX.XX.XX.XX but asterisk 13 writes local ip to the from header. any suggestions? Best Regards, Madushan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an Asterisk 1.4.19.1 server that is behind a Fortinet firewall. Localnet is 192.168.2.0/255.255.255.0 and all external sip devices look as if they are on the same local network because the Fortinet rewrites the incoming IP as its own address. The problem I have is that when I set "externip=148.XXX.XXX.XXX" it is being ignored and I can see SDP packets that have the internal

ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: <sip:test at ekiga.net>;tag=as64618445 To: <sip:test at

Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do

Does anyone know if the 'externip=' in sip.conf is resolved just once at startup or on an on going basis? I would like to use a DNS name through one of the dynamic DNS providers, but if the DNS updates, and asterisk continues using the old resolved value, this could get tricky. Thanks, Dennis