Looks like you have not setup the default nameserver in your machine. In most linux distributions you can edit the /etc/resolv.conf file to setup one. Or use the distro's network manager. Just check if you can resolve the domain using nslookup. Otherwise you can setup pjsip's resolver to do this for you which does not rely on the OS's resolver. sundar On Tue, Feb 28, 2012 at 12:11 PM, Curt Sampson <cjs at cynic.net> wrote: > I'm using PJSIP 2.0-beta (2011/12/29). In one fairly simple application > I have an endpoint that sets up a TLS transport and then attempts to > make a call to an arbitrary SIP(S) URI. > > When I make a call to a SIPS URI with a host part that resolves to an > IP address, albeit one with nobody listening for SIP connections on it, > the program fails in the appropriate manner. However, when I make a call > with a SIPS URI that does not resolve in DNS (i.e., there's no A or any > other type of record for it), the program instead dies due to an ABRT > (6) signal, which appears to be due to the following assertion failure: > > ../src/pj/os_core_unix.c:1201: pj_mutex_lock: Assertion `mutex' failed. > > The logging I get to this point (at level five) is: > > 15:34:40.773 pjsua_call.c Making call with acc #0 to > sips:mixer-1 at bank-a.atp.cynic.net > 15:34:40.773 dlg0x9f4acf4 .UAC dialog created > 15:34:40.773 dlg0x9f4acf4 ..Session count inc to 1 by mod-pjsua > 15:34:40.773 pjsua_media.c .Call 0: initializing media.. > 15:34:40.773 pjsua_media.c ..RTP socket reachable at 192.168.5.34:40000 > 15:34:40.773 pjsua_media.c ..RTCP socket reachable at 192.168.5.34:40001 > 15:34:40.773 pjsua_media.c ..Media index 0 selected for audio call 0 > 15:34:40.773 dlg0x9f4acf4 ..Session count dec to 1 by mod-pjsua > 15:34:40.773 dlg0x9f4acf4 .Module mod-invite added as dialog usage, > data=0x9f4b50c > 15:34:40.773 dlg0x9f4acf4 ..Session count inc to 3 by mod-invite > 15:34:40.773 dlg0x9f4acf4 .Module mod-100rel added as dialog usage, > data=0x9f505f4 > 15:34:40.773 dlg0x9f4acf4 .100rel module attached > 15:34:40.773 inv0x9f4acf4 .UAC invite session created for dialog > dlg0x9f4acf4 > 15:34:40.773 endpoint .Request msg INVITE/cseq=28626 > (tdta0x9f506a0) created. > 15:34:40.773 inv0x9f4acf4 ..Sending Request msg INVITE/cseq=28626 > (tdta0x9f506a0) > 15:34:40.773 dlg0x9f4acf4 ...Sending Request msg INVITE/cseq=28626 > (tdta0x9f506a0) > 15:34:40.773 tsx0x9f526b4 ....Transaction created for Request msg INV > ITE/cseq=28625 (tdta0x9f506a0) > 15:34:40.773 tsx0x9f526b4 ...Sending Request msg INVITE/cseq=28625 > (tdta0x9f506a0) in state Null > 15:34:40.773 sip_resolve.c ....DNS resolver not available, target ' > bank-a.atp.cynic.net:0' type=TLS will be resolved with getaddrinfo() > 15:34:40.828 sip_resolve.c ....Failed to resolve 'bank-a.atp.cynic.net'. > Err=70018 (gethostbyname() has returned error (PJ_ERESOLVE)) > 15:34:40.828 tsx0x9f526b4 ....Failed to send Request msg > INVITE/cseq=28625 (tdta0x9f506a0)! err=70018 (gethostbyname() has returned > error (PJ_ERESOLVE)) > 15:34:40.828 tsx0x9f526b4 ....State changed from Null to Terminated, > event=TRANSPORT_ERROR > 15:34:40.828 dlg0x9f4acf4 .....Transaction tsx0x9f526b4 state changed > to Terminated > Call 0 State: DISCONNCTD > 15:34:40.828 pjsua_media.c .......Call 0: deinitializing media.. > 15:34:40.828 pjsua_media.c .........Media session call00:0 is destroyed > 15:34:40.828 dlg0x9f4acf4 .......Session count dec to 3 by mod-invite > 15:34:40.828 pjsua_call.c .Unable to send initial INVITE request: > gethostbyname() has returned error (PJ_ERESOLVE) [status=70018] > Call 0 State: > 15:34:40.828 dlg0x9f4acf4 .Dialog destroyed > 15:34:40.828 pjsua_media.c .Call 0: deinitializing media.. > > Looks like a bug in PJSIP, given that it's an assertion failure. Anybody > have a fix, or ideas on where this is going wrong, or even thoughts on > how I might debug this? > > cjs > -- > Curt Sampson <cjs at cynic.net> +81 90 7737 2974 > http://www.starling-software.com/ > I have always wished for my computer to be as easy to use as my telephone; > my wish has come true because I can no longer figure out how to use my > telephone. --Bjarne Stroustrup > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20120228/3be18349/attachment-0001.html>
