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 at macro-StdExten:6] Dial("IAX2/barneveld-2036",
"SIP/RickEndpoint&SIP/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 at 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
--
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 <cassius at 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 at macro-StdExten:6] Dial("IAX2/barneveld-2036", > "SIP/RickEndpoint&SIP/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 at 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 > > -- > > > > > > -- > _____________________________________________________________________ > -- 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 >
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 at macro-StdExten:6] Dial("IAX2/barneveld-2036", > "SIP/RickEndpoint&SIP/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 at 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