wiki.VoIP.co.uk

products:asterisk

Chris Harding — Thu, 04 Mar 2010 13:58:49 +0000

These are possibly 'belt and braces' values but they have been working successfully for a considerable amount of time on a live server: PEER Details: authuser= canreinvite=no fromdomain=proxy.voip.co.uk fromuser= host=proxy.voip.co.uk insecure=very password= qualify=yes secret= type=peer username= User Context:

sip:tests

Theo Zourzouvillys — Sun, 07 Jun 2009 04:48:14 +0000

I do a fair bit of SIP interoperability testing and testing other UA devices for compliance before offering them to customers. As such, i'm slowly building up a list of edge cases that are often overlooked when developers write SIP stacks and TU's.

api:sepapi:start - created

Theo Zourzouvillys — Tue, 02 Jun 2009 01:51:43 +0000

HTTP APIs are exported that allow interaction and notifications with any SIP event packages - for example the reg event package, dialog info, message-waiting, rtcp-stats etc. In order to subscribe to a dialog, firstly post a HTTP request to the API controller:

api:aor - created

Theo Zourzouvillys — Tue, 02 Jun 2009 01:04:36 +0000

When not using our user provisioning API, AORs can be processed on dynamic basis. When a request is initially made for an AOR within a configured authority (either through REGISTER or a request which requests the resource), a HTTP callback is made for the AOR's configurating.

api:start - created

Theo Zourzouvillys — Sun, 03 May 2009 03:46:46 +0000

We have a number of RESTful APIs that can be used for programatically configuring your account, performing call-time operations, etc. Available Services: * Account management * Account information * Billing * CDRs * Invoices * Topup Account

sip:case - created

Theo Zourzouvillys — Thu, 05 Feb 2009 16:34:19 +0000

Element Case Sensitive? Via branch insensitive (token) Call-ID sensitive - Byte for byte compare SIP-Version insensitive - must send uppercase though Header field names insensitive From RFC 3261, sect 7.3.1: When comparing header fields, field names are always case-insensitive. Unless otherwise stated in the definition of a particular header field, field values, parameter names, and parameter values are case-insensitive. Tokens are always case-insensitive. Unless specifie…

products:linksys

Theo Zourzouvillys — Sat, 20 Dec 2008 00:54:46 +0000

* PAP2 * SPA200/SPA2002 * Linksys firmware doesn't escape the display-name value. If you have anything except token (a-z, 0-9, and -.!%*_+`'~) or LWS (\r\n\t, and space), then ensure you escape it yourself using '”'.

products:draytek

Theo Zourzouvillys — Fri, 19 Dec 2008 10:45:16 +0000

Draytek 2600/2600x-series - General Notes Calls Dropping after 20 seconds If you find that calls on your local SIP phones drop after approximately 20 seconds, TELNET into the router and issue the following command: sys sip_alg 0 [ENTER] (-- that's a zero on the end) Draytek 2600V-series FXS port setup This ADSL router has two Analogue (FXS) phone ports on the back. To have them successfully connect they need to have different network ports assigned to them, and - very importantly - if you a…

sip:t.38 - created

Theo Zourzouvillys — Fri, 19 Dec 2008 08:00:25 +0000

Example T.38 offer sent by a Cisco AS5350XM on detecting FAX: v=0 o=CiscoSystemsSIP-GW-UserAgent 1107 1281 IN IP4 192.168.0.1 s=SIP Call c=IN IP4 192.168.0.1 t=0 0 m=image 33884 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy

sip:start

Theo Zourzouvillys — Fri, 19 Dec 2008 07:59:32 +0000

* SIP Protocol Frequently Asked Questions * Some comments on digest authentication. * Underspecified situations in the protocol * GRUU notes * Status Codes we send back to SIP UAs * Role of the To header in SIP requests. * Some SIP protocol and stack tests * T.38 examples

products:cisco

Theo Zourzouvillys — Fri, 19 Dec 2008 04:44:03 +0000

To disable the SIP ALG, add these configuration settings: no ip nat service sip tcp port 5060 no ip nat service sip udp port 5060

products:start

Theo Zourzouvillys — Fri, 19 Dec 2008 04:43:10 +0000

See the page on configuration before configuring your SIP device to use VoIP.co.uk. There are a list of requirements your SIP device must meet in order for it to function with VoIP.co.uk. SIP Enabled PBX Systems * Avaya * Cisco Hard Phones * Snom * Grandstream * Linksys * Avaya * Cisco * Mitel * Safecom

ipv6

Theo Zourzouvillys — Sat, 06 Dec 2008 15:06:08 +0000

VoIP.co.uk are dedicated to providing all services provided over IPv4 over IPv6. This includes our SIP platform, web interfaces, APIs, and tertiary services such as SMTP, DNS, etc. DNS Both a.ns.voip.co.uk and b.ns.voip.co.uk have IPv6 support, and have added v6 glue for them.