Re: [asterisk-users] Call does not pass through
Thanks Sam. Please see below CLI log: /[root@localhost ~]# asterisk -r Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer marks...@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@avaya-internal:1] Answer(OOH323/(null)-b7798910, ) in ne w stack -- Executing [s@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@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@avaya-internal:2] Dial(OOH323/(null)-b7798910, SIP/15707088788@cordia) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 15707088788@cordia -- SIP/cordia-0017 answered OOH323/(null)-b7798910 == Spawn extension (avaya-internal, 15707088788, 2) exited non-zero on 'OOH323 /(null)-b7798910' -- Executing [s@avaya-internal:1] Answer(OOH323/(null)-0a389388, ) in ne w stack -- Executing [s@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@avaya-internal:3] WaitExten(OOH323/(null)-0a389388, ) in new stack == CDR updated on OOH323/(null)-0a389388 -- Executing [18772281023@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@avaya-internal:2] Dial(OOH323/(null)-0a389388, SIP/18772281023@cordia) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 18772281023@cordia -- SIP/cordia-0018 is making progress passing it to OOH323/(null)-0a3893 88 -- SIP/cordia-0018 is ringing -- SIP/cordia-0018 is making progress passing it to OOH323/(null)-0a3893 88 -- SIP/cordia-0018 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@localhost ~]# nano /etc/asterisk/extensions_custom.conf [root@localhost ~]# asterisk -r Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer marks...@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@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@192.168.254.15;tag=as4f38e456 To: sip:15707088788@66.148.120.167 Contact: sip:1105@192.168.254.15 Call-ID: 1a2d18961fc1c50a50ecad427e9f350c@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
Re: [asterisk-users] Call does not pass through
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 mr...@mail.altcladding.com.ph wrote: Thanks Sam. Please see below CLI log: *[root@localhost ~]# asterisk -r Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com marks...@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@avaya-internal:1] Answer(OOH323/(null)-b7798910, ) in ne w stack -- Executing [s@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@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@avaya-internal:2] Dial(OOH323/(null)-b7798910, SIP/15707088788@cordia) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 15707088788@cordia -- SIP/cordia-0017 answered OOH323/(null)-b7798910 == Spawn extension (avaya-internal, 15707088788, 2) exited non-zero on 'OOH323 /(null)-b7798910' -- Executing [s@avaya-internal:1] Answer(OOH323/(null)-0a389388, ) in ne w stack -- Executing [s@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@avaya-internal:3] WaitExten(OOH323/(null)-0a389388, ) in new stack == CDR updated on OOH323/(null)-0a389388 -- Executing [18772281023@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@avaya-internal:2] Dial(OOH323/(null)-0a389388, SIP/18772281023@cordia) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 18772281023@cordia -- SIP/cordia-0018 is making progress passing it to OOH323/(null)-0a3893 88 -- SIP/cordia-0018 is ringing -- SIP/cordia-0018 is making progress passing it to OOH323/(null)-0a3893 88 -- SIP/cordia-0018 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@localhost ~]# nano /etc/asterisk/extensions_custom.conf [root@localhost ~]# asterisk -r Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com marks...@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
Re: [asterisk-users] Call does not pass through
Thanks All. Here is my config:
*On my Firewall NAT:*
/I allowed the following ports: 4569,5004-5082, 1-2/
*
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(${RECORDED_FILE})
exten = 9000,7,Wait(2)
exten = 9000,8,Hangup
exten = _,1,Dial(SIP/${EXTEN}@Avaya)
exten = _11XX,1,Dial(H323/${EXTEN}@Avaya)
exten = _X,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten = _X,2,Dial(SIP/${EXTEN}@cordia)
exten = _,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten = _,2,Dial(SIP/${EXTEN}@cordia)/
Regards,
Malvin
On 9/26/2011 9:56 PM, Ruben Rögels wrote:
Am 26.09.2011 13:18, schrieb Malvin Rito:
Hi list,
My call does not pass through on the first dial, I have to redial again
the number for the call to pass through. I'm not sure if the problem is
on my voip proovider or my asterisk server setup. Any thoughts pls?
Regards,
Malvin
Hi,
could be a NAT related issue.
Please be more specific about your setup.
best regards,
Ruben
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Re: [asterisk-users] Call does not pass through
I see a couple of conflicting extensions as well as something I assume
copy-paste malfunction. Please paste the CLI logs of the call.
On Wed, Sep 28, 2011 at 8:26 AM, Malvin Rito
mr...@mail.altcladding.com.phwrote:
Thanks All. Here is my config:
*On my Firewall NAT:*
*I allowed the following ports: 4569,5004-5082, 1-2*
*
On Asterisk Box:*
Here is the extensions.conf:
*[general]
static=yes
autofallthrough=yes
[avaya-internal]
exten = s,1,Answer()
exten = s,2,background(pls-entr-num-uwish2-call)
exten = s,3,WaitExten()
exten = s,4,Dial(SIP/${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 = _,1,Dial(SIP/${EXTEN}@Avaya)
exten = _11XX,1,Dial(H323/${EXTEN}@Avaya)
exten = _X,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten = _X,2,Dial(SIP/${EXTEN}@cordia)
exten = _,1,Authenticate(/etc/asterisk/passcode.txt,a)
exten = _,2,Dial(SIP/${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
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[asterisk-users] Call does not pass through
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 -- _ -- 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
Re: [asterisk-users] Call does not pass through
Can you please post the relevant parts of extensions.conf? As well as a CLI output of when you dial and it fails? On 9/26/11, Malvin Rito mr...@mail.altcladding.com.ph wrote: 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 -- _ -- 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
