search for: ast_gethostbyname

...ion. - app_macro: Remove deprecated module. - res_monitor: Remove deprecated module. - http.c: Minor simplification to HTTP status output. - app_osplookup: Remove obsolete sample config. - say.c: Fix French time playback. (#42) - core: Cleanup gerrit and JIRA references. (#58) - utils.h: Deprecate `ast_gethostbyname()`. (#79) - res_pjsip_pubsub: Add new pubsub module capabilities. (#82) - app_sla: Migrate SLA applications out of app_meetme. - rest-api: Ran make ari stubs to fix resource_endpoints inconsistency - .github: Update AsteriskReleaser for security releases - users.conf: Deprecate users.conf configura...

...ion. - app_macro: Remove deprecated module. - res_monitor: Remove deprecated module. - http.c: Minor simplification to HTTP status output. - app_osplookup: Remove obsolete sample config. - say.c: Fix French time playback. (#42) - core: Cleanup gerrit and JIRA references. (#58) - utils.h: Deprecate `ast_gethostbyname()`. (#79) - res_pjsip_pubsub: Add new pubsub module capabilities. (#82) - app_sla: Migrate SLA applications out of app_meetme. - rest-api: Ran make ari stubs to fix resource_endpoints inconsistency - .github: Update AsteriskReleaser for security releases - users.conf: Deprecate users.conf configura...

...setpwent' in statically linked applications requires at runtime the shared libraries from the glibc version used for linking warning: Using 'endpwent' in statically linked applications requires at runtime the shared libraries from the glibc version used for linking utils.o: In function `ast_gethostbyname': Thanks Cricket on your mind? Visit the ultimate cricket website. Enter http://beta.cricket.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/2c014afd/attachment.htm

Does anyone know if the 'externip=' in sip.conf is resolved just once at startup or on an on going basis? I would like to use a DNS name through one of the dynamic DNS providers, but if the DNS updates, and asterisk continues using the old resolved value, this could get tricky. Thanks, Dennis

Hi All, I had an unusual problem today; I'm sure it's a configuration problem. I had 2 phones behind a nat device and I had qualify=300 in both extensions config. The device I was talking to was an edgewater traffic shaper,/Sip Proxy. Since it is acting as a sip proxy, it was ignoring the OPTIONS messages that * was sending, and thus * interpreted that as the extensions being down. I

Hi, I'm testing Asterisk with a new provider. On calls to US toll-free numbers, there is no audio (calls to normal numbers are ok). In response to a valid INVITE from Asterisk, something like this is received: SIP/2.0 183 Session Progress v:SIP/2.0/UDP [my public IP]:5060;branch=z9hG4bK62d91cea CSeq:103 INVITE i:7a1791cf52d6f3dc2d12b208051d0a21@[provider].com f:"Test User"