Re: [asterisk-users] no progress indication
On 2/18/11 5:18 PM, Paul Belanger pabelan...@digium.com wrote: On 11-02-18 03:59 PM, Cassius Smith wrote: I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). *CLI core show application Progress() CLI output: -- Executing [s@macro-StdExten:6] Dial(IAX2/barneveld-2036, SIP/RickEndpointSIP/xlite-RickEndpoint,20) in new stack == Using SIP RTP CoS mark 5 [Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot connect [Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument -- Called RickEndpoint [Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument == Spawn extension (macro-StdExten, s, 6) exited non-zero on 'IAX2/barneveld-2036' in macro 'StdExten' == Spawn extension (no911, RickEndpoint, 1) exited non-zero on 'IAX2/barneveld-2036' -- Hungup 'IAX2/barneveld-2036' [Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. There is something going wrong here, netsock2 is not parsing the IP address correctly. Are you using realtime? It would be good to see a full debug[1] log of your call. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Hi Paul, no, not using realtime. I collected the trace but it didn't seem to give much clue (at least to me). Here is an extract from the log (dialing extension 4511 this time). Let me know if you want the full debug log including IAX and SIP debugs. (trunk is IAX, endpoints are SIP). [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Macro' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [4511@no911:1] Macro(IAX2/barneveld-9539, StdExten,SIP/4511SIP/xlite-4511,20) in new stack [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'MACRO_EXTEN' is '4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Verbose' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:1] Verbose(IAX2/barneveld-9539, 2,Processing StdExten call for 4511) in new stack [Feb 20 00:23:23] VERBOSE[9962] app_verbose.c: == Processing StdExten call for 4511 [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Verbose [Feb 20 00:23:23] DEBUG[9962] pbx.c: Function result is 'Cassius Home 3703' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Verbose' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:2] Verbose(IAX2/barneveld-9539, 2,CallerID = Cassius Home 3703) in new stack [Feb 20 00:23:23] VERBOSE[9962] app_verbose.c: == CallerID = Cassius Home 3703 [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Verbose [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'ARG1' is 'SIP/4511SIP/xlite-4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Set' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:3] Set(IAX2/barneveld-9539, Device=SIP/4511SIP/xlite-4511) in new stack [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Set [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'MACRO_EXTEN' is '4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Set' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:4] Set(IAX2/barneveld-9539, UserID=4511) in new stack [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Set [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'ARG1' is 'SIP/4511SIP/xlite-4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'ARG2' is '20' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Dial' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:5] Dial(IAX2/barneveld-9539, SIP/4511SIP/xlite-4511,20) in new stack [Feb 20
[asterisk-users] no progress indication
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as expected).
CLI output:
-- Executing [s@macro-StdExten:6] Dial(IAX2/barneveld-2036,
SIP/RickEndpointSIP/xlite-RickEndpoint,20) in new stack
== Using SIP RTP CoS mark 5
[Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot connect
[Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
-- Called RickEndpoint
[Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
[Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
== Spawn extension (macro-StdExten, s, 6) exited non-zero on
'IAX2/barneveld-2036' in macro 'StdExten'
== Spawn extension (no911, RickEndpoint, 1) exited non-zero on
'IAX2/barneveld-2036'
-- Hungup 'IAX2/barneveld-2036'
[Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102 (Critical
Request) -- See doc/sip-retransmit.txt.
Here is my StdExten macro:
[macro-StdExten]
exten = s,1,Verbose(2,Processing StdExten call for
${MACRO_EXTEN})
exten = s,n,Verbose(2,CallerID = ${CALLERID(all)})
exten = s,n,Set(Device=${ARG1})
exten = s,n,Set(UserID=${MACRO_EXTEN})
exten = s,n,Dial(${ARG1},${ARG2})
exten = s,n,Verbose(2,== Voicemail ${MACRO_EXTEN} -- unavail)
exten = s,n,Voicemail(${MACRO_EXTEN}@default,u)
exten = s,n,Hangup()
I was expecting the system to indicate that ringing was ?
I know I can check channel availability to bypass this behavior; just
curious why it's happening or whether it's expected.
Cassius
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Re: [asterisk-users] no progress indication
Try to use Answer() in your dial plan. Not sure though but it had been
resoved my issue years ago.
--
Sent from my iPhone
On Feb 18, 2011, at 3:59 PM, Cassius Smith cass...@cassius.org wrote:
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with
VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't
get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as expected).
CLI output:
-- Executing [s@macro-StdExten:6] Dial(IAX2/barneveld-2036,
SIP/RickEndpointSIP/xlite-RickEndpoint,20) in new stack
== Using SIP RTP CoS mark 5
[Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot
connect
[Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit:
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid
argument
-- Called RickEndpoint
[Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full:
Unable to
create channel of type 'SIP' (cause 20 - Unknown)
[Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit:
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid
argument
[Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit:
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid
argument
[Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit:
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid
argument
== Spawn extension (macro-StdExten, s, 6) exited non-zero on
'IAX2/barneveld-2036' in macro 'StdExten'
== Spawn extension (no911, RickEndpoint, 1) exited non-zero on
'IAX2/barneveld-2036'
-- Hungup 'IAX2/barneveld-2036'
[Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit:
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid
argument
[Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit:
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid
argument
[Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit:
sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid
argument
[Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102
(Critical
Request) -- See doc/sip-retransmit.txt.
Here is my StdExten macro:
[macro-StdExten]
exten = s,1,Verbose(2,Processing StdExten call for
${MACRO_EXTEN})
exten = s,n,Verbose(2,CallerID = ${CALLERID(all)})
exten = s,n,Set(Device=${ARG1})
exten = s,n,Set(UserID=${MACRO_EXTEN})
exten = s,n,Dial(${ARG1},${ARG2})
exten = s,n,Verbose(2,== Voicemail ${MACRO_EXTEN} -- unavail)
exten = s,n,Voicemail(${MACRO_EXTEN}@default,u)
exten = s,n,Hangup()
I was expecting the system to indicate that ringing was ?
I know I can check channel availability to bypass this behavior; just
curious why it's happening or whether it's expected.
Cassius
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Re: [asterisk-users] no progress indication
On 11-02-18 03:59 PM, Cassius Smith wrote: I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). *CLI core show application Progress() CLI output: -- Executing [s@macro-StdExten:6] Dial(IAX2/barneveld-2036, SIP/RickEndpointSIP/xlite-RickEndpoint,20) in new stack == Using SIP RTP CoS mark 5 [Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot connect [Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument -- Called RickEndpoint [Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument == Spawn extension (macro-StdExten, s, 6) exited non-zero on 'IAX2/barneveld-2036' in macro 'StdExten' == Spawn extension (no911, RickEndpoint, 1) exited non-zero on 'IAX2/barneveld-2036' -- Hungup 'IAX2/barneveld-2036' [Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. There is something going wrong here, netsock2 is not parsing the IP address correctly. Are you using realtime? It would be good to see a full debug[1] log of your call. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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
