呼叫不成功

-- Executing [13764487625@ss7:1] Set("SIP/8660-08f86290", "CALLERID(all)=13258292653#0707") in new stack
[Mar 10 00:50:58] DEBUG[6963]: pbx.c:1842 pbx_extension_helper: Launching 'Dial'
-- Executing [13764487625@ss7:2] Dial("SIP/8660-08f86290", "SS7/10157#13764487625") in new stack
[Mar 10 00:50:58] DEBUG[6963]: l4isup.c:824 ss7_requester: SS7 request (SS7/10157#13764487625) format = 0x8.
[Mar 10 00:50:58] DEBUG[6963]: l4isup.c:867 ss7_requester: SS7 channel SS7/10157#13764487625 allocated successfully.
[Mar 10 00:50:58] DEBUG[6963]: rtp.c:1584 ast_rtp_make_compatible: Channel 'SS7/|s1/94' has no RTP, not doing anything
[Mar 10 00:50:58] DEBUG[6963]: channel.c:3323 ast_channel_inherit_variables: Not copying variable SIPCALLID.
[Mar 10 00:50:58] DEBUG[6963]: channel.c:3323 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT.
[Mar 10 00:50:58] DEBUG[6963]: channel.c:3323 ast_channel_inherit_variables: Not copying variable SIPDOMAIN.
[Mar 10 00:50:58] DEBUG[6963]: channel.c:3323 ast_channel_inherit_variables: Not copying variable SIPURI.
[Mar 10 00:50:58] DEBUG[6963]: l4isup.c:1998 ss7_call: SS7 call, addr=10157#13764487625, cid=13258292653#0707(0x0/Presentation Allowed, Not Screened) CIC=94. linkset '|s1'
[Mar 10 00:50:58] DEBUG[6963]: l4isup.c:1867 isup_send_iam: chan_ss7: isup_send_iam: ISDN_H324M is not set.
[Mar 10 00:50:58] DEBUG[6963]: l4isup.c:413 mtp_enqueue_isup_packet: Queue packet CIC=94, len=42, linkset='|s1', link='|2', slinkset='|s1', slink='|1'
-- Sent IAM CIC=94 ANI=13258292653#0707 DNI=10157#13764487625 RNI=
-- Called 10157#13764487625
[Mar 10 00:50:58] DEBUG[6963]: channel.c:2808 set_format: Set channel SS7/|s1/94 to read format ulaw
[Mar 10 00:50:58] DEBUG[6963]: channel.c:2808 set_format: Set channel SIP/8660-08f86290 to read format alaw
[Mar 10 00:50:58] DEBUG[6936]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 8660-08f86290
[Mar 10 00:50:58] DEBUG[6941]: mtp.c:1944 mtp_thread_main: Queue MSU, lsi=0, last_send_ix=0, linkset=|s1, m->link=|1
[Mar 10 00:50:58] DEBUG[6936]: chan_sip.c:16004 sip_devicestate: Checking device state for peer 8660-08f86290
[Mar 10 00:50:58] DEBUG[6941]: mtp.c:1652 mtp2_fill_zaptel_buf: Sending buffer to zaptel len=46, on link '|1' bsn=15, fsn=19.
[Mar 10 00:50:58] DEBUG[6963]: chan_sip.c:6751 transmit_response_with_sdp: Setting framing from config on incoming call
[Mar 10 00:50:58] DEBUG[6963]: chan_sip.c:6515 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True
[Mar 10 00:50:58] DEBUG[6963]: chan_sip.c:6516 add_sdp: ** Our prefcodec: 0x0 (nothing)
[Mar 10 00:50:58] DEBUG[6963]: chan_sip.c:6647 add_sdp: -- Done with adding codecs to SDP
[Mar 10 00:50:58] DEBUG[6963]: channel.c:2342 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=85)
[Mar 10 00:50:58] DEBUG[6963]: chan_sip.c:6692 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw)
[Mar 10 00:50:58] DEBUG[6963]: rtp.c:2769 ast_rtp_write: Ooh, format changed from unknown to ulaw
[Mar 10 00:50:58] DEBUG[6963]: rtp.c:2786 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160
[Mar 10 00:50:58] DEBUG[6936]: devicestate.c:287 do_state_change: Changing state for SIP/8660-08f86290 - state 1 (Not in use)
[Mar 10 00:50:58] DEBUG[6936]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 8660
[Mar 10 00:50:58] DEBUG[6936]: chan_sip.c:16004 sip_devicestate: Checking device state for peer 8660
[Mar 10 00:50:58] DEBUG[6936]: devicestate.c:287 do_state_change: Changing state for SIP/8660 - state 1 (Not in use)
[Mar 10 00:50:58] DEBUG[6946]: app_queue.c:659 handle_statechange: Device 'SIP/8660-08f86290' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 00:50:58] DEBUG[6946]: app_queue.c:659 handle_statechange: Device 'SIP/8660' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Mar 10 00:50:58] DEBUG[6942]: chan_sip.c:4620 find_call: = Found Their Call ID: c104fc11b400f63fNTM3ZjRhZGQwNTI1NDE0NzY4M2E2NTg1NGQ0YTIzNTE. Their Tag 723e8254 Our tag: as39a5ae0c
[Mar 10 00:50:58] DEBUG[6942]: chan_sip.c:15341 handle_request: **** Received INVITE (5) - Command in SIP INVITE
[Mar 10 00:50:58] DEBUG[6942]: chan_sip.c:15365 handle_request: Ignoring SIP message because of retransmit (INVITE Seqno 2, ours 2)
[Mar 10 00:50:58] DEBUG[6963]: rtp.c:874 ast_rtcp_read: RTCP NAT: Got RTCP from other end. Now sending to address 58.246.60.78:33127
[Mar 10 00:50:58] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 132 bytes
[Mar 10 00:50:58] DEBUG[6963]: rtp.c:1180 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 58.246.60.78:33126
[Mar 10 00:51:01] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 176 bytes
[Mar 10 00:51:04] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 176 bytes
[Mar 10 00:51:07] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 176 bytes
[Mar 10 00:51:10] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 176 bytes
[Mar 10 00:51:10] DEBUG[6942]: chan_sip.c:2096 __sip_autodestruct: Auto destroying SIP dialog '85300807e154100dNTM3ZjRhZGQwNTI1NDE0NzY4M2E2NTg1NGQ0YTIzNTE.'
[Mar 10 00:51:10] DEBUG[6942]: chan_sip.c:3319 sip_destroy: Destroying SIP dialog 85300807e154100dNTM3ZjRhZGQwNTI1NDE0NzY4M2E2NTg1NGQ0YTIzNTE.
Really destroying SIP dialog '85300807e154100dNTM3ZjRhZGQwNTI1NDE0NzY4M2E2NTg1NGQ0YTIzNTE.' Method: ACK
[Mar 10 00:51:12] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 176 bytes
[Mar 10 00:51:14] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 176 bytes
[Mar 10 00:51:17] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 176 bytes
[Mar 10 00:51:20] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 176 bytes
[Mar 10 00:51:23] DEBUG[6963]: rtp.c:879 ast_rtcp_read: Got RTCP report of 176 bytes
[Mar 10 00:51:23] NOTICE[6941]: l4isup.c:1053 t7_timeout: T7 timeout (waiting for ACM or CON) CIC=94.
[Mar 10 00:51:23] DEBUG[6941]: channel.c:1384 ast_softhangup_nolock: Soft-Hanging up channel 'SS7/|s1/94'
[Mar 10 00:51:23] DEBUG[6963]: channel.c:1483 ast_hangup: Hanging up channel 'SS7/|s1/94'
-- SS7 hangup 'SS7/|s1/94' CIC=94 Cause=16 (state=2)
[Mar 10 00:51:23] DEBUG[6963]: l4isup.c:2070 ss7_hangup: SS7 hangup 'SS7/|s1/94' CIC=94 (state=2), chan=0x08f87820
[Mar 10 00:51:23] DEBUG[6963]: l4isup.c:2101 ss7_hangup: SS7 hangup 'SS7/|s1/94' CIC=94 cause=16
[Mar 10 00:51:23] DEBUG[6963]: l4isup.c:413 mtp_enqueue_isup_packet: Queue packet CIC=94, len=16, linkset='|s1', link='|2', slinkset='|s1', slink='|1'
[Mar 10 00:51:23] DEBUG[6963]: devicestate.c:304 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SS7/|s1/94
-- No one is available to answer at this time (1:0/0/0)
[Mar 10 00:51:23] DEBUG[6963]: rtp.c:1509 ast_rtp_early_bridge: Channel '<unspecified>' has no RTP, not doing anything
[Mar 10 00:51:23] DEBUG[6963]: app_dial.c:1734 dial_exec_full: Exiting with DIALSTATUS=NOANSWER.
[Mar 10 00:51:23] DEBUG[6963]: pbx.c:1842 pbx_extension_helper: Launching 'Hangup'
-- Executing [13764487625@ss7:3] Hangup("SIP/8660-08f86290", "") in new stack

问一下在没改chan_ss7之前有没有这个呼不通的问题？（关于号码中间有#号的）

在没有修改#号键，也没有呼通。

10157#后面不要呼移动的号码，呼一个联通的手机号测试一下。看结果如何。

感觉这个问题跟Asterisk和Chan_ss7关系不大了。

hi:
如果要通话，你必须首先确认你的chan_ss7 是通的吗？ 是 up 的， inservice ? 如果确认是通的，你再做呼叫，直接这样测试：
exten=> 100,1,dial(chan_ss7/linkset/10000) 或者呼叫手机。

现在呼叫成功了，但是相互不能通话，出现ss7_write: Write buffer full on CIC=94 (wrote only 0 of 160), audio lost (suppress 5)报错，请问是什么原因呢！谢谢！