hugolivude
2006-Apr-30 18:36 UTC
[Asterisk-Users] WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'
Hi, Red Hat 9.0 Asterisk 1.2.7.1 Whenever I start Asterisk, I am unable to call out on my SIP channel:>-- Executing Dial("Zap/1-1", "SIP/6137451576@6477235412||t|") in new stack >Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No suchhost: 6477235412>Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create >channel of type 'SIP' (cause 3 - No route to destination)However, if I RELOAD my dial plan from the CLI I get this message, it starts to work. I think I've tracked it down to the following warning message:>Apr 30 11:01:21 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unableto lookup '????' This message doesn't reappear when I do a RELOAD. Anyone know what this is all about? My SIP.conf is below. Notice how I've commented out auth=md5. This seems to have eliminated the following WARNING message that used appear just before the one above:>Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822add_realm_authentication: Format for >authentication entry is user[:secret]@realm at line 31 Line 31 of my sip.conf was auth=md5. I was able to get the SIP channel working with this warning as well, but it took a lot more RELOADs. Any ideas? SIP.conf ===== [general] ; context=incoming-bogus-calls bindport=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) maxexpirey=3600 ; Must be larger than the re-register timeout on the router defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; register=>6477235412:<mypassword>@sip.unlimitel.ca/6477235412 externip=<mystaticIPaddress> ;Outside address localnet=192.168.0.148/255.255.255.0 ;Inside Network ; ;******************************************************************** [6477235412] type=peer ;auth=md5 username=6477235412 fromuser=6477235412 fromdomain=unlimitel.ca secret=<mypassword> host=sip.unlimitel.ca port=5060 nat=yes canreinvite=no qualify=no disallow=all allow=g729 dtmfmode=rfc2833 insecure=very context=incoming ; ;---------------------------------------------------------------------
hugolivude
2006-May-01 04:33 UTC
[Asterisk-Users] WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'
Hi, Red Hat 9.0 Asterisk 1.2.7.1 Whenever I start Asterisk, I am unable to call out on my SIP channel:>-- Executing Dial("Zap/1-1", "SIP/6137451576@6477235412||t|") in new stack >Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No suchhost: 6477235412>Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create >channel of type 'SIP' (cause 3 - No route to destination)However, if I RELOAD my dial plan from the CLI I get this message, it starts to work. I think I've tracked it down to the following warning message:>Apr 30 11:01:21 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unableto lookup '????' This message doesn't reappear when I do a RELOAD. Anyone know what this is all about? My SIP.conf is below. Notice how I've commented out auth=md5. This seems to have eliminated the following WARNING message that used appear just before the one above:>Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822add_realm_authentication: Format for >authentication entry is user[:secret]@realm at line 31 Line 31 of my sip.conf was auth=md5. I was able to get the SIP channel working with this warning as well, but it took a lot more RELOADs. Any ideas on how I can get my SIP channel working properly, without these warning messages & w/o having to do a RELOAD each time? SIP.conf ===== [general] ; context=incoming-bogus-calls bindport=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) maxexpirey=3600 ; Must be larger than the re-register timeout on the router defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; register=>6477235412:<mypassword>@sip.unlimitel.ca/6477235412 externip=<mystaticIPaddress> ;Outside address localnet=192.168.0.148/255.255.255.0 ;Inside Network ; ;******************************************************************** [6477235412] type=peer ;auth=md5 username=6477235412 fromuser=6477235412 fromdomain=unlimitel.ca secret=<mypassword> host=sip.unlimitel.ca port=5060 nat=yes canreinvite=no qualify=no disallow=all allow=g729 dtmfmode=rfc2833 insecure=very context=incoming ; ;---------------------------------------------------------------------
hugolivude
2006-May-01 06:10 UTC
[Asterisk-Users] WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'
Hi, Red Hat 9.0 Asterisk 1.2.7.1 ** Apologies if you notice this posted multiple times, I'm just not seeing it on the boards ** Whenever I start Asterisk, I am unable to call out on my SIP channel:>-- Executing Dial("Zap/1-1", "SIP/6137451576@6477235412||t|") in new stack >Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No suchhost: 6477235412>Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create >channel of type 'SIP' (cause 3 - No route to destination)However, if I RELOAD my dial plan from the CLI I get this message, it starts to work. I think I've tracked it down to the following warning message:>Apr 30 11:01:21 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unableto lookup '????' This message doesn't reappear when I do a RELOAD. Anyone know what this is all about? My SIP.conf is below. Notice how I've commented out auth=md5. This seems to have eliminated the following WARNING message that used appear just before the one above:>Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822add_realm_authentication: Format for >authentication entry is user[:secret]@realm at line 31 Line 31 of my sip.conf was auth=md5. I was able to get the SIP channel working with this warning as well, but it took a lot more RELOADs. Any ideas on how I can get my SIP channel working properly, without these warning messages & w/o having to do a RELOAD each time? SIP.conf ===== [general] ; context=incoming-bogus-calls bindport=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) maxexpirey=3600 ; Must be larger than the re-register timeout on the router defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; register=>6477235412:<mypassword>@sip.unlimitel.ca/6477235412 externip=<mystaticIPaddress> ;Outside address localnet=192.168.0.148/255.255.255.0 ;Inside Network ; ;******************************************************************** [6477235412] type=peer ;auth=md5 username=6477235412 fromuser=6477235412 fromdomain=unlimitel.ca secret=<mypassword> host=sip.unlimitel.ca port=5060 nat=yes canreinvite=no qualify=no disallow=all allow=g729 dtmfmode=rfc2833 insecure=very context=incoming ; ;---------------------------------------------------------------------
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