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[asterisk2billing.org] INSTALL :: RE: Direct Dialling out on SIP/IAX Trunks: However I have reached an enpass, after loading the database with suppliers, trunks, ratecards, rates and users I cannot get the system to dial out directly on a sip or iax trunk. I have shown below the relative contexts and .

[Untitled] ２３．RT-200NE（PR-200NE)対応（５）extension.conf: exten => s,n,GotoIf($[$["${DIALSTATUS}"="CHANUNAVAIL"]|$["${DIALSTATUS}"="CONGESTION"]]?continue:hangup)

[Untitled] All calls are being treated as national | trixbox: We have finally worked out (with much appreciated assistance from Sangoma) that all calls are being placed as NATIONAL with the number exactly as dialed. I'm not sure that's clear so I'll give an example: I would normally expect to be able to dial "9201234" (local) or "01519201234" (national) or "001519201234" (international) and get connected to 0151 920 1234 (here in the UK).

[Untitled] Asterisk 1.6: It is worth noting as well the possibility of changing the timing for the SIP protocol, the so-called timers, defined in RFC 4028. This may require, for example, in cases where the network is characterized by stable long delay packages, as well as avoiding the hang SIP sessions, when communication with the SIP client fetches up during the session itself.

[VOIP-info.org Wiki Changes] Asterisk Avoid SIP NAT Traversal: On MacOSX, the SIP/IAX gateway setup for FWD as shown in the above diagram can be very easily configured using the Asterisk Assistants for MacOSX. It takes only a couple of minutes including installation and it requires no prior knowledge of Asterisk or VoIP.

[Untitled] IAX2 Trunk between 2 Trixboxes | trixbox: IAX is not an Internet Standard and you are over complicating the scenario. The definitive docs are at asterisk.org and voip-info.org.

[Untitled] Incoming DTMF issues - Teliax Forums: A new problem arose today with our asterisk voicemail system, using Teliax provisioned numbers and colocated at the Teliax facility, that has never occurred before. We beleive it has something to do with the most recent Teliax updates.

[Untitled] PHP_AGI学习笔记_外星人麦客: 使用phpagi程序可以非常容易的实现IVR菜单，外部数据库访问，呼叫控制。等功能。 首先准备环境，测试帐号两个sip帐号和一个iax2帐号. sip.conf.

[Untitled] Henri Cook » Blog Archive » Installing Asterisk on Ubuntu Server ...: If the hardphone has been configured proprely and made all the appropriate connections (which wasn’t hard for me, just set the usernames and secrets for Ph0 in ‘account 1′ and the same server address for incoming/outbound and you’re on the way (do this after you’ve configured the server).

[Untitled] IP PBX | VoIP PBX |PABX: 7 Get Under The Bonnet 51. 7.1 Editing The .conf Files. 52. 7.1.1 iax.conf 52. 7.1.2 Indications.conf 52. 7.1.3 enum.conf 52. 7.1.4 extensions.conf 52. 7.1.5 extensions_custom.conf 53. 7.1.6 sip.conf 53. 7.2 Port Forwarding - Routers. ...

[Untitled] Archives - SipTheeSkype SIP Skype Gateway Update 20081101: The question is, first step, 300 call through 200 till 100, it works out just fine, by not terminating step one, I continue on step two which error occurs, when I try to call out from 600 to 500 till 400, the connection between step two just breaks down after accepting (picking up the phone) the call, and at the terminal node 600(which is winXP) it shows an error message on the skype client screen which says “ error in terminal audio device ” , at the same time in step one, the SIP phone(100) occurs error, it can hear the voice from the other side, but the other side can’t here form it, which means the speaker of the sip phone(100) is dead, any thing you say using/from 100 are not available.

[Untitled] Nerd Vittles » Newbie’s Guide to Asterisk@Home 2.4: Unabridged ...: We need a consistent IP address or domain name both on your internal network and externally if you expect to receive incoming calls reliably. There are three pieces to the IP configuration: (1) setting the internal IP address of your Asterisk server, (2) configuring an external qualified domain name which will always point to your router/firewall, and (3) configuring your router to transfer incoming Asterisk packets to your Asterisk server.

[Untitled] IAX and SIP: There have been some things that I skipped over because I didn’t think anyone would want to read a 50 page blog post about SIP and IAX. If you have any questions or if I didn’t cover something that needs covering, leave a comment and let me know.

[Untitled] Bobcares Blog » Asterisk -A quick look under the hood: sip.conf and/or iax.conf. These generally depend on your carrier and the connection type that you choose with them.

Reflected tags on Technorati: Blog, Voicemail Conf, VoxxMail