[asterisk-users] Call does not pass through
Sam Govind
govoiper at gmail.com
Wed Sep 28 01:17:22 CDT 2011
Hey,
So far the Dialplan execution is ok, despite the conflicts and some other
mistakes like repeating priorities in it but they're not involved in this
call.
You'r A-leg is H323 endpoint and Destination is on SIP. I'm now thinking
about codec mismatch on first try Tell me this happens every time? Like
first call fails for sure and second call goes through?
if so please post the tcpdump/wireshark traces for a failed call as well as
successful call separately. Also NAting could be the reason like Rube
suspects - but the call should be failing everytime. Anyways post the
complete h323 as well as SIP traces combined for each failed & successful
call.
Regards.
-Sammy
On Wed, Sep 28, 2011 at 10:59 AM, Malvin Rito <mrito at mail.altcladding.com.ph
> wrote:
> Thanks Sam. Please see below CLI log:
>
> *[root at localhost ~]# asterisk -rvvvv
> Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
> Created by Mark Spencer <markster at digium.com> <markster at digium.com>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
> detail
>
> s.
> This is free software, with components licensed under the GNU General
> Public
> License version 2 and other licenses; you are welcome to redistribute it
> under
> certain conditions. Type 'core show license' for details.
> =========================================================================
> == Parsing '/etc/asterisk/asterisk.conf': == Found
> Connected to Asterisk 1.6.2.7 currently running on localhost (pid = 11533)
> Verbosity is at least 4
> == Spawn extension (avaya-internal, 15707088788, 2) exited non-zero on
> 'OOH323
>
> /(null)-b7798910'
> -- Executing [s at avaya-internal:1] Answer("OOH323/(null)-b7798910", "")
> in ne
>
> w stack
> -- Executing [s at avaya-internal:2] BackGround("OOH323/(null)-b7798910",
> "pls-
>
> entr-num-uwish2-call") in new stack
> -- <OOH323/(null)-b7798910> Playing 'pls-entr-num-uwish2-call.gsm'
> (language
>
> 'en')
> == CDR updated on OOH323/(null)-b7798910
> -- Executing [15707088788 at avaya-internal:1]
> Authenticate("OOH323/(null)-b779
>
>
> 8910", "/etc/asterisk/passcode.txt,a") in new stack
> -- <OOH323/(null)-b7798910> Playing 'agent-pass.ulaw' (language 'en')
> -- <OOH323/(null)-b7798910> Playing 'auth-thankyou.ulaw' (language
> 'en')
> -- Executing [15707088788 at avaya-internal:2]
> Dial("OOH323/(null)-b7798910", "
>
>
> SIP/15707088788 at cordia") in new stack
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Called 15707088788 at cordia
> -- SIP/cordia-00000017 answered OOH323/(null)-b7798910
> == Spawn extension (avaya-internal, 15707088788, 2) exited non-zero on
> 'OOH323
>
> /(null)-b7798910'
> -- Executing [s at avaya-internal:1] Answer("OOH323/(null)-0a389388", "")
> in ne
>
> w stack
> -- Executing [s at avaya-internal:2] BackGround("OOH323/(null)-0a389388",
> "pls-
>
> entr-num-uwish2-call") in new stack
> -- <OOH323/(null)-0a389388> Playing 'pls-entr-num-uwish2-call.gsm'
> (language
>
> 'en')
> -- Executing [s at avaya-internal:3] WaitExten("OOH323/(null)-0a389388",
> "") in
>
> new stack
> == CDR updated on OOH323/(null)-0a389388
> -- Executing [18772281023 at avaya-internal:1]
> Authenticate("OOH323/(null)-0a38
>
>
> 9388", "/etc/asterisk/passcode.txt,a") in new stack
> -- <OOH323/(null)-0a389388> Playing 'agent-pass.ulaw' (language 'en')
> -- <OOH323/(null)-0a389388> Playing 'auth-thankyou.ulaw' (language
> 'en')
> -- Executing [18772281023 at avaya-internal:2]
> Dial("OOH323/(null)-0a389388", "
>
>
> SIP/18772281023 at cordia") in new stack
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Called 18772281023 at cordia
> -- SIP/cordia-00000018 is making progress passing it to
> OOH323/(null)-0a3893
>
> 88
> -- SIP/cordia-00000018 is ringing
> -- SIP/cordia-00000018 is making progress passing it to
> OOH323/(null)-0a3893
>
> 88
> -- SIP/cordia-00000018 answered OOH323/(null)-0a389388
> == Spawn extension (avaya-internal, 18772281023, 2) exited non-zero on
> 'OOH323
>
> /(null)-0a389388'
> == Manager 'admin' logged on from 127.0.0.1
> == Manager 'admin' logged off from 127.0.0.1
> == Manager 'admin' logged on from 127.0.0.1
> == Manager 'admin' logged off from 127.0.0.1
> localhost*CLI>
> Disconnected from Asterisk server
> Executing last minute cleanups
> [root at localhost ~]# nano /etc/asterisk/extensions_custom.conf
> [root at localhost ~]# asterisk -rvvvv
> Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
> Created by Mark Spencer <markster at digium.com> <markster at digium.com>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
> details.
> This is free software, with components licensed under the GNU General
> Public
> License version 2 and other licenses; you are welcome to redistribute it
> under
> certain conditions. Type 'core show license' for details.
> =========================================================================
> == Parsing '/etc/asterisk/asterisk.conf': == Found
> Connected to Asterisk 1.6.2.7 currently running on localhost (pid = 11533)
> Verbosity is at least 4
> -- Remote UNIX connection
> localhost*CLI> sip set debug peer cordia
> SIP Debugging Enabled for IP: 66.148.120.167:5060
> localhost*CLI> core set verbose 0
> Verbosity is now OFF
> Audio is at 192.168.254.15 port 19144
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Reliably Transmitting (no NAT) to 66.148.120.167:5060:
> INVITE sip:15707088788 at 66.148.120.167 SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.15:5060;branch=z9hG4bK1e0698f8;rport
> Max-Forwards: 70
> From: "10.1.129.247" <sip:1105 at 192.168.254.15>;tag=as4f38e456
> To: <sip:15707088788 at 66.148.120.167>
> Contact: <sip:1105 at 192.168.254.15>
> Call-ID: 1a2d18961fc1c50a50ecad427e9f350c at 192.168.254.15
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.2.7
> Date: Wed, 28 Sep 2011 02:43:24 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 2082067001 2082067001 IN IP4 192.168.254.15
> s=Asterisk PBX 1.6.2.7
> c=IN IP4 192.168.254.15
> t=0 0
> m=audio 19144 RTP/AVP 0 8
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> localhost*CLI>
> <--- SIP read from UDP:66.148.120.167:5060 --->
> SIP/2.0 100 Trying
> CSeq: 102 INVITE
> Via: SIP/2.0/UDP 222.127.178.113:5060;branch=z9hG4bK1e0698f8;rport
> From: "10.1.129.247" <sip:1105 at 192.168.254.15>;tag=as4f38e456
> Call-ID: 1a2d18961fc1c50a50ecad427e9f350c at 192.168.254.15
> To: <sip:15707088788 at 66.148.120.167>;tag=28091911111014129494936357
> Contact: <sip:66.148.120.167:5060;transport=udp>
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> localhost*CLI>
> <--- SIP read from UDP:66.148.120.167:5060 --->
> SIP/2.0 200 OK
> CSeq: 102 INVITE
> Via: SIP/2.0/UDP 222.127.178.113:5060;branch=z9hG4bK1e0698f8;rport
> From: "10.1.129.247" <sip:1105 at 192.168.254.15>;tag=as4f38e456
> Call-ID: 1a2d18961fc1c50a50ecad427e9f350c at 192.168.254.15
> To: <sip:15707088788 at 66.148.120.167>;tag=28091911111014129494936357
> Contact: <sip:66.148.120.167:5060;transport=udp>
> Content-Type: application/sdp
> Content-Length: 225
>
> v=0
> o=VoipSwitch 6356 7356 IN IP4 66.148.120.167
> s=VoipSIP
> i=Audio Session
> c=IN IP4 66.148.120.167
> t=0 0
> m=audio 6356 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
>
> <------------->
> --- (9 headers 11 lines) ---
> Found RTP audio format 0
> Found RTP audio format 101
> Found audio description format PCMU for ID 0
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
> (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1
> (telephone-event), combined - 0x0 (nothing)
> Peer audio RTP is at port 66.148.120.167:6356
> list_route: hop: <sip:66.148.120.167:5060;transport=udp>
> set_destination: Parsing <sip:66.148.120.167:5060;transport=udp> for
> address/port to send to
> set_destination: set destination to 66.148.120.167, port 5060
> Transmitting (no NAT) to 66.148.120.167:5060:
> ACK sip:66.148.120.167:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.15:5060;branch=z9hG4bK28aa746f;rport
> Max-Forwards: 70
> From: "10.1.129.247" <sip:1105 at 192.168.254.15>;tag=as4f38e456
> To: <sip:15707088788 at 66.148.120.167>;tag=28091911111014129494936357
> Contact: <sip:1105 at 192.168.254.15>
> Call-ID: 1a2d18961fc1c50a50ecad427e9f350c at 192.168.254.15
> CSeq: 102 ACK
> User-Agent: Asterisk PBX 1.6.2.7
> Content-Length: 0
>
>
> ---
> localhost*CLI>*
>
> Regards,
> Malvin
>
>
> On 9/28/2011 1:46 PM, Sam Govind wrote:
>
> I see a couple of conflicting extensions as well as something I assume
> copy-paste malfunction. Please paste the CLI logs of the call.
>
> On Wed, Sep 28, 2011 at 8:26 AM, Malvin Rito <
> mrito at mail.altcladding.com.ph> wrote:
>
>> Thanks All. Here is my config:
>>
>> *On my Firewall NAT:*
>>
>> *I allowed the following ports: 4569,5004-5082, 10000-20000*
>> *
>> On Asterisk Box:*
>>
>> Here is the extensions.conf:
>> *[general]
>> static=yes
>> autofallthrough=yes
>>
>> [avaya-internal]
>> exten => s,1,Answer()
>> exten => s,2,background(pls-entr-num-uwish2-call)
>> exten => s,3,WaitExten()
>> exten => s,4,Dial(SIP/${EXTEN})
>> exten => s,5,Dial(H323/${EXTEN})
>> exten => s,6,PlayBack(vm-nobodyavail)
>> exten => s,7,HangUp()
>>
>> exten => 1000,1,Dial(SIP/1000)
>> exten => 1000,1,Answer()
>>
>> exten => 1000,2,PlayBack(vm-goodbye)
>> exten => 1000,3,HangUp()
>>
>> #Extension for recording
>> exten => 9000,1,Answer()
>> exten => 9000,2,Background(pm-to-record-phrase)
>> exten => 9000,3,Hangup()
>> #exten => 9000,3,Wait(2)
>> exten => 9000,4,Record(alt_recording%d:ulaw)
>> exten => 9000,5,Wait(2)
>> exten => 9000,6,Playback(${RECORDED_FILE})
>> exten => 9000,7,Wait(2)
>> exten => 9000,8,Hangup
>>
>> exten => _XXXX,1,Dial(SIP/${EXTEN}@Avaya)
>> exten => _11XX,1,Dial(H323/${EXTEN}@Avaya)
>>
>> exten => _XXXXXXXXXXXXX,1,Authenticate(/etc/asterisk/passcode.txt,a)
>> exten => _XXXXXXXXXXXXX,2,Dial(SIP/${EXTEN}@cordia)
>>
>> exten => _XXXXXXXXXXXX,1,Authenticate(/etc/asterisk/passcode.txt,a)
>> exten => _XXXXXXXXXXXX,2,Dial(SIP/${EXTEN}@cordia)*
>>
>>
>>
>> Regards,
>> Malvin
>>
>>
>> On 9/26/2011 9:56 PM, Ruben Rögels wrote:
>>
>> Am 26.09.2011 13:18, schrieb Malvin Rito:
>>
>> Hi list,
>> My call does not pass through on the first dial, I have to redial again
>> the number for the call to pass through. I'm not sure if the problem is
>> on my voip proovider or my asterisk server setup. Any thoughts pls?
>>
>> Regards,
>> Malvin
>>
>> Hi,
>>
>> could be a NAT related issue.
>>
>> Please be more specific about your setup.
>>
>> best regards,
>> Ruben
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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