Asterisk telephone landing of several ways, ET263 set

PSTN fixed: Personal, PC plugged into the analog-to-digital conversion card, such as the one-port digium x100p,4 mouth TDM 400, Taobao on its own find.

GSM Phone: Bluetooth access, requires a desktop, a Bluetooth receiver, configuration Chan_mobile

Digital E1 Line: Cable/digital cable provided by telecommunication companies, usually used by company users, such as Digium Digital Board card

SIP service: Foreign bee Vbuzzer, domestic ET263 China line service

Skype: Requires Skye connect integration

GSM Gateway: To more than 800 RMB a hardware, the GSM SIM card transferred to SIP ======================================================================== ========================

Skype Connect

Be harmonious http://www.skype.com/zh-Hans/features/skype-connect/

1) Use your office phone to receive Skype calls

2) Enjoy a cheap call by integrating Skype into your existing phone system.

By adding Skype Connect to an existing SIP-enabled PBX, your company can save on communication costs with minimal upgrades or no additional upgrades.

How to use: Sign in to Skype Manager to purchase a channel and assign points to your SIP profile. Configure the PBX or VoIP gateway as detailed in the SIP profile. Start calling

Start using Skype Connect now.

================================================================================================ ET263 is set successfully, Console always error: warning[3247]: chan_sip.c:24433 handle_incoming:misrouted SIP response ' 401 Unauthorized ' with Call-id ' 05d2126 24f8017f1114dde8441d6ebdc@xxx.homeftp.org ', too many vias

It's chan_sip.c in/usr/src/asterisk-1.8.10.1/channels.
Put this phrase: if (!ast_strlen_zero (__get_header (req, "via", &via_pos)) {
Ast_log (log_warning, "misrouted SIP response '%s ' with Call-id '%s ', too many vias\n", E, callid);
return 0;
}
The copy code becomes:/* if (!ast_strlen_zero (__get_header (req, "via", &via_pos)) {
Ast_log (log_warning, "misrouted SIP response '%s ' with Call-id '%s ', too many vias\n", E, callid);
return 0;
}*/
Duplicate codes

Then compile, then make it. Windows Xlite can be paged out via et 263.

Reference:

Http://www.telecom-cafe.com/forum/viewthread.php?tid=4110&extra=page%3D1&page=1

================================================================================================

Now exhale with BlackBerry sip phone, the error on the asterisk console is as follows:

] warning[8829]: channel.c:5799 ast_channel_make_compatible_helper:no path to translate from sip/myet263_out-00000005 To sip/101-00000004
= = Spawn Extension (macro-dialout-trunk, S,) exited Non-zero on ' sip/101-00000004 ' in macro ' dialout-trunk '

Check core show translation found that the SIP Trunk myet263_out definition inside the g729 and AMR Voice codec can not translate.

Workaround:

Modify the definition of SIP Trunk myet263_out in the sip_additional.conf file

allow=g729,ulaw,alaw,gsm,g726
Modify to allow=ulaw,alaw,gsm,g726

Reboot asterisk can be from blackberry sip phone, walk et 263 call my mobile number/fixed.

But at home, not in the office, both sides have no voice, well, this should be a NAT problem, need to install a stun server. Maybe 3CX Phone system can help.

To test: After returning to the office, test BlackBerry SIP call out.