Firewall, NAT y SIP

Sip ,trunking, nat y los firewall...los cuales no son faciles.. (http://www.ingate.com/)

Ingate SIParators

http://www.ingate.com/SIP-trunking.php

he most common solution for enterprises with a local IP-PBX to connect to the PSTN is via a local PSTN gateway. However, today more and more service providers offer one single connection for both data and voice between the carrier and the enterprise. This is commonly called a SIP trunk. For the enterprise this means that they no longer need either the local PSTN gateway or costly PRI/BRI?s as the local IP-PBX can be connected to the service provider?s PSTN gateways over the Internet.

The SIP Trunking software module, working in conjunction with an Ingate Firewall or SIParator, solves the Network Address Translation (NAT) traversal issues that are faced by businesses using a SIP trunk. Together, they control both incoming and outgoing communications and route the communication to the intended users. All voice traffic (as well as data traffic) must traverse the enterprise firewall/NAT. However, SIP traffic cannot traverse traditional enterprise firewalls and NAT devices. As a result, the firewall/NAT device blocks all SIP traffic, which includes VoIP. Ingate resolves this issue, enabling enterprises to utilize SIP trunks and enjoy the benefits of SIP trunking while maintaining the security and integrity of their network.
A cost-efficient solution, the SIP trunk reduces monthly expenses for the enterprise as they:

* Only need one connection for both data and voice
* No longer need PRI/BRI connections
* Don?t have to buy or maintain a local PSTN-gateway

Firewalls and Benefits of Proxy-Based Solutions -- Technical details

The Ingate Firewall® or Ingate SIParator® solves the basic problem of firewall and NAT traversal using a SIP proxy-based implementation. The SIP proxy is a complete solution to the firewall and NAT traversal issues introduced by the enterprise firewall. A proxy is designed to briefly stop the signaling packets so that each one can be inspected before the header information is rewritten. The SIP proxy also makes sure the packets are delivered to the appropriate endpoints. This provides the enterprise with a flexible, controlled implementation of SIP-based communications that also maintains voice/video quality. Ingate Firewalls and SIParators offer a number of advantages:

* Near-end NAT traversal for connecting SIP communications between an enterprise and a carrier
* Dynamically open and close the media ports to admit voice or video on both UDP and TCP ports
* Encryption for communication privacy and prevention of eavesdropping. Include termination, transcoding and passthrough of TLS and SRTP
* Authentication and registration at the carrier side
* Give priority of voice traffic using the Ingate Quality of Service features
* ENUM look-up to route the call IP-IP all the way if possible
* Least cost routing of international calls

Confirmed interoperability

Ingate has confirmed interoperability several of the leading IP-PBX vendors and ITSPs. See the full list here.
Startup Tool for SIP Trunking

The Startup Tool is an installation tool for Ingate Firewall® and Ingate SIParator® products using the Ingate SIP Trunking software module, which facilitates the setup of complete SIP trunking solutions. The tool will automatically configure a user?s Ingate Firewall or SIParator to work with the IP-PBX or SIP trunking service provider of their choice. With the push of a button, the configuration tool will automatically create a SIP trunk connection designed to the user?s individual setup.
Read more about the tool here.

Read more about the Ingate SIP Trunking software module

Learn more about the Ingate Firewall

Learn more about the Ingate SIParator

For additional information on Ingate SIP trunking solution please contact info@ingate.com

DISA - Direct Inward System Access

DISA ()
Direct Inward System Access leaves out the people who call into the system and provides them with an internal dial tone.
DISA (password [context [callerid [mailbox [@-voice context ]]]])
DISA (file with a password, [callerid [mailbox [@-voice context]]]) provides a dial tone internal callers outside so that they can make calls as if calling from inside extension . Upon hearing the dial tone, an access code must be entered followed by the "#". If it is correct, the caller hears another tone, which is the system of dial tone and the caller can make and initiate calls.
[Caution] This type of access represents a serious threat to security and real and must be carefully considered and planned before use, but you should be used at all!
The option is a numerical code that access should be introduced to the caller to make calls. Following this syntax, all users on the floor using the same access code. If you want to allow access unsecured, enter the string "no password," instead of a real password.
The context option specifies the context in which it began the call is placed. If not provided, DISA () assumes that the context called disabilities.
The callerid option determines the number of mail (and the optional voice context) of a mailbox. If the mailbox containing new messages to the caller will hear a stuttered dial tone to indicate this.
Alternatively, you can use the password file to define multiple access passwords. Each line of this file can contain either an access code or a combination of an access code and a context, separated by the "|" character (pipe). If not specified the context, it is assumed disability.
If the caller is authenticated successfully with a valid access code, DISA () will start the call in the context specified.

WebDial con Asterisk

http://voipbox.libits.org/?l=en&p=webdial#s

Te puede servir como ejemplo....bueno estuvimos hablando del uso de BASH...
No se si es bien lo que necesitas...pero puede servir.

Un abrazo ,Helius

Webdial with Asterisk
For Webdial to work with Asterisk, a web server that can handle CGI scripts, has to run on the same machine. When you click the OK button of the input box, a small popup window is created, in which the CGI script is run. It first closes the popup, then creates a call file that rings your phone and when you pick up the receiver, Asterisk connects you to the number you clicked on. Therefore it needs to know your extension number in addition to the number you want to dial. A sample shell script to achieve this is provided below, but this can be just as easily coded in Perl or PHP. It uses the query arguments, channel, from for the caller, to for the number to call, context and timeout to determine how many seconds to ring your phone. Some changes to the sample might be on order to make it work in your environment.

#!/bin/sh
echo -e "Content-type: text/html\\n\\n" # Close the popup
callfile=/tmp/`sed -e 's/[^0-9]//g' /proc/uptime`.call # Create a unique filename
eval `echo $QUERY_STRING|sed -e 's/&/ /g'` # Create variables from arguments
cat <$callfile # See here for callfile syntax
Channel: $channel/$from
Callerid: $from
WaitTime: $timeout
Context: $context
Priority: 1
Extension: $to
EOF
mv $callfile /var/spool/asterisk/outgoing


Make sure the script is executable for the user, your web server is running under. To test it enter something like
http://server/path/webdial.cgi?channel=SIP&context=default&from=123&to=4567890&timeout=15
in your browser, where channel is the channel your phone is connected to, context is the one for calling out, from is your extension, to is a number you want to dial and timeout the number of seconds, you want your phone to ring. If this works, you can create the bookmarklet.
Create Webdial bookmarklet to work with Asterisk

Helius Fax

Javier mira estos emails cortos..sobre Sangoma, Siemens, faxesHay muchas variables a ver, ademas de los problemas reales con los aparatos de fax (falta de papel, roturas, etc) y las limitantes actuales entre trixbox y la Siemens. Por eso creo, que seria muy importante tener un ambiente de pruebas..!!Y no pasar las penurias en el cliente como hoy...ademas de la falta de posibilidades por el tipo de conexion. ------------------------------Siememes HiPath 3550------------------------------------------------------No son los mismos valores que para Uruguay ojo!!Muy parecido a lo egregamos hoy al DryTek..!Estava tendo alguns problemas com uma Siememes HiPath 3550. Coloquei no indications.conf [br]description = Brazilringcadence = 1000,4000dial = 425busy = 425/250,0/250ring = 425/1000,0/4000congestion = 425/250,0/250,425/750,0/250callwaiting = 425/50,0/1000 ; Dialrecall not used in Brazil standard (using UK standard) dialrecall = 350+440; Record tone is not used in Brazil, use busy tonerecord = 425/250,0/250; Info not used in Brazil standard (using UK standard) info = 950/330,1400/330,1800/330stutter = 350+440 zaptel.conf defaultzone=brloadzone=br zapata.conflanguage=br sip.conf [general] country=br E resolvi o problema. ---------------------------Fax through Sangoma------------------------------------------------------- in our company we are trying to do this: Fax <--> Traditional PBX <--> Asterisk <--> PSTN In practice, we have put an Asterisk equipped with a Sangoma A102 (2 PRI ports) between our PBX ( Siemens HiCom) and the PSTN in order to have a VoIP network along the traditional telephony network. The problem is with the fax. We just want to send and receive faxes from/to our fax machine connected to the Siemens (without needing any interaction with our VoIP network, the faxes are sent to/received from PSTN). Unfortunately we are experiencing a lot of problems: the faxes not always work and when they work, it's likely to have incomplete pages. I know that faxing with VoIP is very troublesome, but maybe someone else is using a similar configuration and he found a good configuration or maybe has some hints to improve the results. We are using Asterisk 1.2.13.-------------------------Sangoma will detect the tones and disable echo cancel---------------------Incoming faxes, the Sangoma will detect the tones and disable echo cancel. To send outbound, you will have to add another trunk group, of one or more channels and disable echo cancellation and use that to dial out. Example (/etc/asterisk/zapata.conf)blah blahechocancel=yesblah blah group = 1channel =>1-20 blah blahecho cancel=nogroup = 2channel=22-23 So you would use for faxing specifically Zap/g2/ and it will use channel 22 or 23 but with echo turned off. Use Zap/g1 and it will use the first group of channels with echo cancel on (or whatever other parameters come before the group command)----------------------------------- ULAW ou ALAW ------------------------------------------------------Renato, FAX sobre VoIP é um negócio chato, nunca funciona 100%.Pense na possibilidade de colocar uma Placa FXS direto no seu Asterisk.DIGIVOICE -> ASTERISK -> FXS -> Aparelho de FAX.Assim pode ligar o FAX direto no Asterisk e não precisa converter sua chamada para VoIP e jogar na rede para o ATA, pois a latência,perda de pacotes e outros motivos influenciam diretamente no envio de FAX.Considerações,FAX e Multifuncionais modernas enviam FAX na velocidade de 33.6Kbps, enviando através de ATA / VoIP não vai funcionar.veja se consegue abaixar a velocidade no seu aparelho.Ative o recurso de correção de erro do seu fax (Se tiver)Lembre-se de sempre usar ULAW ou ALAW para envio de fax, se for outro codec com maior compressão não funciona.Certifique-se que está fixando o codec no sip.conf.Experimente desligar o cancelamento de echo do seu ATA, normalmente ele faz isso automaticamente ao detectar o sinal de fax, masé necessário forçar-----------------------------------T.38 support in Asterisk------------------------------------------The T.38 support in Asterisk 1.4 is only pass through. Asterisk is notan endpoint or gateway of any sort. It only works passthrough inSIP to SIP calls without chan_local or chan_agent involved. Both endpoints of the call will need T.38 support and have T.38enabled in sip.conf.Asterisk 1.4 as I see it is still a bit unstable and not ready forproduction use. Be careful out there.-------------------------------------------------------------------------------------------------------------------Podria seguir pegando Emails de Asterisk, trixbox y Siemens...pero no es el caso...Solo queria enviarte los que considere importantes..!! --------------------------------------------------------------------------------------------------------------------- "Nós somos o que fazemos no dia-a-dia. A excelência, portanto não é um ato, mas sim um hábito" - Aristóteles Grupo de Usuarios y Desarrolladores Debian del Uruguay Debian Uruguay (Guddú)http://debianuruguay.org/Debian (デビアン) は、あなたのコンピュータのためのフリーなオペレーティングシステム (OS) です しかし、OS の基本ツールのほとんどが GNU プロジェクトに由来しているため、GNU/Linux と名付けています。---

Helius tonos antel

Me imagino que esto debe ser cosa de cada dia, por ser una empresa de telefonia...
Por las dudas, se los envios.
Tonos Uruguay
http://www.3amsystems.com/wireline/tone-search.htm?start=0&kTone=1004
Country Tone Freq/Hz Cadence/s
Uruguay Busy tone 425 0.5 on 0.5 off
Uruguay Congestion tone 425 0.25 on 0.25 off
Uruguay Dial tone - I 425 continuous
Uruguay Dial tone - II 450 0.7 on 0.8 off 0.2 on 0.3 off
Uruguay Negative indication tone 425 0.5 on 0.5 off
Uruguay Positive indication tone 425 2x(0.1 on 0.2 off) 0.1 on 1.3 off
Uruguay Ringing tone 425 1.0 on 4.0 off
Uruguay Special information tone 950/1400/1800 3x0.333 on 1.0 off
Uruguay Waiting tone 425 0.2 on 0.2 off 0.2 on 4.4 off
Uruguay Warning tone - operator intervening 425 0.15 on 0.25 off 0.15 on 1.45 off
Ademas tiene ejemplos de configuracion:
Switch to a different display format: [Sipura/Linksys] [Zaptel/Asterisk]

Ejemplo Country Tone:
Sipura/Linksys
Uruguay Busy tone 425@xx;yy(0.5/0.5/1)
Uruguay Congestion tone 425@xx;yy(0.25/0.25/1)
Uruguay Dial tone - I 425@xx;yy(*/0/1)
Uruguay Dial tone - II 450@xx;yy(0.7/0.8/1,0.2/0.3/1)
Uruguay Negative indication tone 425@xx;yy(0.5/0.5/1)
Uruguay Positive indication tone 425@xx;yy(0.1/0.2/1,0.1/0.2/1,0.1/1.3/1)
Uruguay Ringing tone 425@xx;yy(1/4/1)
Uruguay Special information tone 950@xx,1400@xx,1800@xx;yy(0.333/0/1,0.333/0/2,0.333/1/3)
Uruguay Waiting tone 425@xx;yy(0.2/0.2/1,0.2/4.4/1)
Uruguay Warning tone - operator intervening 425@xx;yy(0.15/0.25/1,0.15/1.45/1)
Country Tone
Zaptel/Asterisk string
Uruguay Busy tone 425/500,0/500
Uruguay Congestion tone 425/250,0/250
Uruguay Dial tone - I 425
Uruguay Dial tone - II 450/700,0/800,450/200,0/300
Uruguay Negative indication tone 425/500,0/500
Uruguay Positive indication tone 425/100,0/200,425/100,0/200,425/100,0/1300
Uruguay Ringing tone 425/1000,0/4000
Uruguay Special information tone 950/333,1400/333,1800/333,0/1000
Uruguay Waiting tone 425/200,0/200,425/200,0/4400
Uruguay Warning tone - operator intervening 425/150,0/250,425/150,0/1450

Helius Fax

por lo de "Pobres condiciones linea"Did you change the rxgain or the rx setting? I changed the rxgain from 0.0 to 20.0. I might make some more test calls, reducing the rxgain to see at what setting the incoming fax detection breaks. Might invoke ztmonitor as well.Are you using the rxfax module or have you changed to nvfaxdetect? This time around I was using the rxfax module only. One variation that I noticed with AAH 1.5 was: (no son Trixbox, pero vienen bien)Without NVFaxDetect-Internal SIP fax -> 666 = OK. Internal SIP fax-> 7777 = Failure (Does not recognise fax & dials my default test ext). Incoming PSTN fax -> AAH 1.5 = OK.With NVFaxDetect-Internal SIP fax -> 666 = OK.Internal SIP fax-> 7777 = OK.Incoming PSTN fax -> AAH 1.5 = OK.Now I understand what you mean by fax detection as distinct from fax handshaking/negotiation.How I sent the test FAXHere is my ztmonitor during sending. (I split the rx and tx onto separate lines to get around the wp formatting) and then during the reception of the Telstra fax report.SENDING A FAXpbx:~# /usr/src/zaptel-1.2.2/ztmonitor 1 -vvVisual Audio Levels.-------------------- Use zapata.conf file to adjust the gains if needed.( # = Audio Level * = Max Audio Hit )<----------------(RX)----------­ ------><----------------(TX)----------­ ------>######################* ########################*########################*########################*####################### *######################## *######################## *######################## *######################## * ######################## *##########################*RECEIVING A FAXpbx:~# /usr/src/zaptel-1.2.2/ztmonitor 1 -vvVisual Audio Levels.--------------------Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit )<----------------(RX)----------­ ------>#########*#########*#########*#########*##########*######### *######### *##########*<----------------(TX)----------­ ------> Telstra FaxStream reported:Transmission level: -17.1dBm EXCELLENTLine noise level: -54.3dBm EXCELLENTTelstra recommends fax transmission levels be of -15.0 to -17.0 dBmother reported scores were all EXCELLENT with 100.00% error free (yay!!)My ztmonitor txgain is about 2/3 to 3/4 up the scale and is a smidge OVER the Telstra recommendation so I could drop it a notch or two.looks like the notion of getting the ztmonitor vu meter about half way is indeed what we need (I like to test these things for myself!)Like I thought before.... I could up my rxgain a bit and reduce the txgain but I found a bit of audible hiss crept into the voice calls with rxgain over 13.I did change my setting to:rxgain=11.0txgain=-3.0-- "Nós somos o que fazemos no dia-a-dia. A excelência, portanto não é um ato, mas sim um hábito" - Aristóteles Grupo de Usuarios y Desarrolladores Debian del Uruguay Debian Uruguay (Guddú)http://debianuruguay.org/Debian (デビアン) は、あなたのコンピュータのためのフリーなオペレーティングシステム (OS) です しかし、OS の基本ツールのほとんどが GNU プロジェクトに由来しているため、GNU/Linux と名付けています。---

Helius Fax

First off, do not use any version over 0.0.3. I am using 0.0.3 on centos 4.5, asterisk 1.2.24 and freepbx 2.3 and it is working very well. One very important thing to keep in mind is that faxing over voip will only work reliably with ulaw or alaw and your internet connection MUST be able to sustain a constant data stream with low jitter. If your interested I have a shell script to install asterisk 1.2.24 and
freepbx-2.3 with rxfax and txfax on centos 4 and working on centos 5.