Hello,
Thanks for your reply. Here are the answers:
-Softphones/hardphones work from other off-sites (although they have different NAT routers (linksys, netgear, dlink, sonicwall, etc)
-In/Out calls to PTSN go through the same SIP trunk to a voip provider
-If it were a codec issue, why do calls from landline-->x-lite work fine at that users location? Only calls from x-lite-->landline produce no audio at that users location (see 1-3 in my original post)
-I've tried "PBX delivers audio" both ways (default is off) and no luck.
Here is the server log NOT working for x-lite-->landline:
I can see that it says no RTP packets were received, the remoteAddr is the off-site location. What would cause this? Especially since landline-->x-lite and x-lite-->voicemail/otherextensions work fine.....?
Code:
09:59:49.131 Active calls counted toward license limit: []
09:59:22.835 [MS105000] C:46.1: No RTP packets were received:remoteAddr=xxx.xxx.140.161:40224,extAddr=yyy.yyy.100.84:9010,localAddr=yyy.yyy.100.84:9010
09:59:22.381 [CM503008]: Call(46): Call is terminated
09:59:17.131 Active calls counted toward license limit: [46]
09:59:15.225 [MS211000] C:46.2: zzz.zzz.19.12:15792 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
09:59:13.335 [CM503007]: Call(46): Device joined: sip:zzz.zzz.19.12:5060
09:59:13.335 [CM503007]: Call(46): Device joined: sip:[email protected]:16438
09:59:13.319 [CM505003]: Provider:[Burloak] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Asterisk PBX] PBX contact: [sip:yyy.yyy.100.84:5060]
09:59:13.319 [CM503002]: Call(46): Alerting sip:zzz.zzz.19.12:5060
09:59:04.475 [CM503025]: Call(46): Calling VoIPline:nnnnnn3895@(Ln.10000@Burloak)@[Dev:sip:zzz.zzz.19.12:5060]
09:59:04.381 [CM503004]: Call(46): Route 1: VoIPline:nnnnnn3895@(Ln.10000@Burloak)@[Dev:sip:zzz.zzz.19.12:5060]
09:59:04.381 [CM503010]: Making route(s) to "nnnnnn3895"<sip:[email protected]>
09:59:04.366 [CM505001]: Ext.223: Device info: Device Identified: [Man: Counterpath;Mod: X-Lite;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [X-Lite release 1104o stamp 56125] PBX contact: [sip:[email protected]:5060]
09:59:04.366 [CM503001]: Call(46): Incoming call from Ext.223 to "nnnnnn3895"<sip:[email protected]>
09:58:45.131 Active calls counted toward license limit: []
Here is the server log WORKING for xlite-->voicemail (999):
Code:
10:40:17.187 Active calls counted toward license limit: []
10:39:51.640 [CM503008]: Call(51): Call is terminated
10:39:45.187 Active calls counted toward license limit: [51]
10:39:40.859 [CM503007]: Call(51): Device joined: sip:[email protected]:40600;rinstance=65aa450e32db0bba
10:39:40.859 [CM503007]: Call(51): Device joined: sip:[email protected]:16438
10:39:40.843 [CM505001]: Ext.999: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] PBX contact: [sip:[email protected]:5060]
10:39:40.843 [CM503002]: Call(51): Alerting sip:[email protected]:40600;rinstance=65aa450e32db0bba
10:39:40.687 [CM503025]: Call(51): Calling Ext:Ext.999@[Dev:sip:[email protected]:40600;rinstance=65aa450e32db0bba]
10:39:40.640 [CM503004]: Call(51): Route 1: Ext:Ext.999@[Dev:sip:[email protected]:40600;rinstance=65aa450e32db0bba]
10:39:40.640 [CM503010]: Making route(s) to "999"<sip:[email protected]>
10:39:40.625 [CM505001]: Ext.223: Device info: Device Identified: [Man: Counterpath;Mod: X-Lite;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [X-Lite release 1104o stamp 56125] PBX contact: [sip:[email protected]:5060]
10:39:40.625 [CM503001]: Call(51): Incoming call from Ext.223 to "999"<sip:[email protected]>
10:39:13.156 Active calls counted toward license limit: []
QUESTIONS:
1) This seems to be a network/NAT issue but if thats the case why do some calls from x-lite work (voice mail, other extension) yet others dont (landline)?
2) Could this be an issue related to the voip provider? (
note: landline-->xlite works fine, and the log above seems to imply no RTP is coming in from the remote site - but why only when dialing a landline? Does x-lite know the difference/treat things differently depending on the destination being dialed?)
3) What is the difference between (landline-->x-lite) which works and (x-lite-->landline) which does not work? Also, (x-lite-->voicemail) and (x-lite-->other extensions) work fine too. How is that possible? Does x-lite treat different destinations differently?
What do I do? Going nuts here...
Thanks