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 -- Probation passed - setting RTP source address to
192.
168.1.138:55542
> 0x1d08efa0 -- Probation passed - setting RTP source address to
217.195.xx.xx:46346
-- SIP/FD-0000010e answered SIP/101-0000010d
> 0x1d08efa0 -- Probation passed - setting RTP source address to
217.195.xx.xx:46346
thanks and regards.
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So you are saying that it resolved the issue to activate voicemail on the device that 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 > > 0x2b393cfc2610 -- Probation passed - setting RTP source address > to 192. > 168.1.138:55542 > > 0x1d08efa0 -- Probation passed - setting RTP source address to > 217.195.xx.xx:46346 > -- SIP/FD-0000010e answered SIP/101-0000010d > > 0x1d08efa0 -- Probation passed - setting RTP source address to > 217.195.xx.xx:46346 > thanks and regards. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150320/a10dae29/attachment.html>
hi the issue still the same i have 2 trunks whe i configure the first in x-lite and the second in my server or my ip-phone snom320 directly from x-lite i can call my trunk without issue but when i try ti call from snom320 to x-lite or from my server asterisk using extension in x-lite the call all time is failed any help please thanks and regards 2015-03-20 19:28 GMT+00:00 Trey Hilyard <kctrey at gmail.com>:> So you are saying that it resolved the issue to activate voicemail on the > device that 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 >> > 0x2b393cfc2610 -- Probation passed - setting RTP source address >> to 192. >> 168.1.138:55542 >> > 0x1d08efa0 -- Probation passed - setting RTP source address to >> 217.195.xx.xx:46346 >> -- SIP/FD-0000010e answered SIP/101-0000010d >> > 0x1d08efa0 -- Probation passed - setting RTP source address to >> 217.195.xx.xx:46346 >> thanks and regards. >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150324/87de1d3c/attachment.html>