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拨打大陆电话用ET263真的不错,一点问题也没有,值得大家考虑!

本帖最後由 角色 於 2011-3-21 12:35 編輯

一般给大陆打电话,用ET263真的不错!清晰度非常高。一般最高都是人民币七分钱一分钟。

角色

ET263 是不是免費注冊的? 我在網頁看了很久也不知道怎樣注冊

TOP

可以到官方网站注册,都是免费,但是拨打大陆电话要0.12元一分钟。

我们一般号码都是从淘宝代理拿的,一般是不要钱,有的要钱,一个号码是收一块钱。


角色

TOP

請問可以用ET263作Sip Acc嗎?

TOP

Add ET263 as usual sip trunk in IP01, then modify two files:
1. In sip.conf add a line to auth.:
register=588xxxxxx:password@sip.etelephone.net:10002/588xxxxxx
2. In user.conf:
registersip=no   ;changed from registersip=yes

Using CLI command sip show peers, which confirmed that ET263 can be reached. However, I cannot dial out thru ET263 as circuit busy was returned.

TOP

Please show me the response if you issue the command "sip show peers".

YH

TOP

本帖最後由 Qnewbie 於 2011-3-24 17:17 編輯

Sip show peers in the web emulator:
Name/username              Host            Dyn Nat ACL Port     Status               
588xxxxxx/588xxxxxx        211.150.115.14       N      10002    OK (344 ms)

Dial rule:
exten=_0086xx.,1,Macro(trunkdial-failover-0.3,${588xxxxxx}/0${EXTEN:4},,588xxxxxx,,${CALLERID(num)},${EXTEN:0})

Debug info from CLI:
[Mar 24 10:00:30] NOTICE[376]: chan_sip.c:13457 handle_response_peerpoke: Peer '588xxxxxx' is now Reachable. (347ms / 2000ms)

   -- Executing [008675512345678@DLPN_internal:1] Macro("SIP/2000-0000000d", "trunkdial-failover-0.3|SIP/588xxxxxx/075512345678||588xxxxxx||2000|008675512345678") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:1] Set("SIP/2000-0000000d", "CALLERID(num)=2000") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:2] Set("SIP/2000-0000000d", "TOUCH_MIXMONITOR=2000-008675512345678-20110324-100436") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/2000-0000000d", "TOUCH_MIXMONITOR_FORMAT=wav49") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:4] GotoIf("SIP/2000-0000000d", "0?1-dial|1") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:5] Set("SIP/2000-0000000d", "CALLERID(all)=") in new stack
    -- Executing [s@macro-trunkdial-failover-0.3:6] Goto("SIP/2000-0000000d", "1-dial|1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/2000-0000000d", "SIP/588xxxxxx/075512345678||") in new stack
    -- Called 588xxxxxx/075512345678
[Mar 24 10:04:37] WARNING[376]: chan_sip.c:13120 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk@xxx.xxx.xxx.xxx>;tag=as2425966a'
    -- SIP/588xxxxxx-0000000e is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/2000-0000000d", "0 > 0 ?1-CONGESTION|1:1-out|1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-out,1)
    -- Executing [1-out@macro-trunkdial-failover-0.3:1] Hangup("SIP/2000-0000000d", "") in new stack
  == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on 'SIP/2000-0000000d' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_internal, 008675512345678, 1) exited non-zero on 'SIP/2000-0000000d'

TOP

You may take a look at the following link which shows the steps to set up an ET263 trunk on IP-01.

http://www.hkepc.com/forum/viewt ... age=296#pid22731855

YH

TOP

I set up the account according to post 5 in this thread and the following:
ET263.png

TOP

How was the results?

YH

TOP

Modify the outbound rule with Dial:
  1. exten=_0086x.,1,Dial(SIP/588xxxxxx/0${EXTEN:4},,r)
  2. exten=_0086x.,n,Hangup()
複製代碼
Got 503, "Service unavailable" back

TOP

我的Asterisk是1.8.3.2,需要打个补丁才能让ET263注册成功,别的VSP没有问题。

TOP

回復 12# bradyzhu


    What is the patch? Would you kindly give us a link or release it in this forum? TIA!

TOP

回復 13# Qnewbie

It's my patch.

I create it based on 1.8.2.3, and I can register to ET263 again with it.

--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -23607,10 +23607,10 @@
                /* RFC 3261 - 8.1.3.3 If more than one Via header field value is present in a reponse
                 * the UAC SHOULD discard the message. This is not perfect, as it will not catch multiple
                 * headers joined with a comma. Fixing that would pretty much involve writing a new parser */
-                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;
-                }
+//                 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;
+//                 }
                if (p->ocseq && (p->ocseq < seqno)) {
                        ast_debug(1, "Ignoring out of order response %d (expecting %d)\n", seqno, p->ocseq);
                        return -1;

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回復 14# bradyzhu

Are you a programmer? You know how to patch.

We do need your help here.

I would like cmphone to be registered in Asterisk and can dial in and out

Register is sometimes ok but so far I can dial out

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