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！

角色

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

bradyzhu

回復  bradyzhu


    Thanks!

One more thing, I am wondering if you can dial out/in with ET263 afte ...
Qnewbie 發表於 2011-5-17 22:54

After patch, I can dial out through ET263. I do not need the incoming from ET263.

My code base is 1.8.2.3, and my patch is based on it. I do not know about the 1.4.x.

The old version asterisk is deleted from OpenWRT during introducing the 1.8.

Qnewbie

Sorry, delete.

Qnewbie

回復 14# bradyzhu


    Thanks!

One more thing, I am wondering if you can dial out/in with ET263 after this patch?

I use currently 437M based on asterisk1.4.3x(?) which cannot apply the patch. The registration with ET263 does work but not dial in/out.

The latest Asterisk1.4.4x(?) can use this patch. However, 506M from bubblestar C-hing and my own restarts frequently in my IP01 for unknown reason

ckleea

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

bradyzhu

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;

Qnewbie

回復 12# bradyzhu


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

bradyzhu

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。

Qnewbie

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

角色

How was the results?

YH

Qnewbie

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

下載 (32.03 KB)
ET263

2011-3-25 16:47

角色

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

Qnewbie

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'

角色

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

YH

Qnewbie

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.