search for: 0033149xxxxxx

i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d > 0x2b393cfc2610 -- Proba...

...sub-record-check:5] Return("SIP/101-00000103", "") in new stack -- Executing [out at sub-record-check:3] Return("SIP/101-00000103", "") in new stack -- Executing [0149xxxxxx at from-internal:5] Macro("SIP/101-00000103", "dialout-trunk,5,0033149xxxxxx,,off") in new stack -- Executing [s at macro-dialout-trunk:1] Set("SIP/101-00000103", "DIAL_TRUNK=5") in new stack -- Executing [s at macro-dialout-trunk:2] GosubIf("SIP/101-00000103", "0?sub-pincheck,s,1()") in new stack -- Executing [s at m...

thanks for your response i noticed that when i active the voicemail in the IP-phone where the number 0033149xxxxxx is configured i can call this number without issue the server asterisk and the ip-phone where the number is configured are in the same network 192.168.1.X Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording SIP/101-0000010d...

I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: "Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060" Are you sure that "0033149xxxxxx" is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but normally a 556 indicates that your provider didn't have routing for either the R-URI or they didn't recognize that is was coming from you. You m...

...t sits past your trunk provider? That confuses me a little, but if your calls are working, that's great news. On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit < salah.elharit200 at gmail.com> wrote: > i noticed that when i active the voicemail in the IP-phone where the > number 0033149xxxxxx is configured i can call this number without issue > > Using SIP RTP TOS bits 184 > == Using SIP RTP CoS mark 5 > -- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording > SIP/101-0000010d > -- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d >...

...en i configure the trunk in my server and i use > > extension > > > > all the ip-phone and x-lite and server asterisk in the same network > > 192.168.1.x > > > > == Using SIP RTP TOS bits 184 > > == Using SIP RTP CoS mark 5 > > -- Called SIP/FD/0033149XXXXXX > > -- SIP/FD-000000b9 is making progress passing it to SIP/306-000000b8 > > > 0x2afec424c430 -- Probation passed - setting RTP source address > to > > 192.168.1.212:57592 > > > 0xc5922b0 -- Probation passed - setting RTP source address to > &gt...

...trunk in x-lite i can call theses ip-phones without issue the problem just when i configure the trunk in my server and i use extension all the ip-phone and x-lite and server asterisk in the same network 192.168.1.x == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149XXXXXX -- SIP/FD-000000b9 is making progress passing it to SIP/306-000000b8 > 0x2afec424c430 -- Probation passed - setting RTP source address to 192.168.1.212:57592 > 0xc5922b0 -- Probation passed - setting RTP source address to 217.195.xx.xxx:29674 -- Got SIP response 556 &quo...

...without issue > the problem just when i configure the trunk in my server and i use > extension > > all the ip-phone and x-lite and server asterisk in the same network > 192.168.1.x > > == Using SIP RTP TOS bits 184 > == Using SIP RTP CoS mark 5 > -- Called SIP/FD/0033149XXXXXX > -- SIP/FD-000000b9 is making progress passing it to SIP/306-000000b8 > > 0x2afec424c430 -- Probation passed - setting RTP source address to > 192.168.1.212:57592 > > 0xc5922b0 -- Probation passed - setting RTP source address to > 217.195.xx.xxx:29674 >...