search for: dialmap

...the trailing # is also going to get in the way of transferring to another extension (or any other dial codes that might need to be used from these analog phones). Is their either a way to strip the # off the end of the dialing so it isn't passed thru, or is there a place I can specify a dialmap for the FXS ports so when my dial pattern is matched it just dials right away (or set a dial timeout as right now there doesn't appear to be one, it just waits forever for you to finish dialing). Or is none of this possible with FXS ports on Sangoma cards and I should look at getting a...

...as excellent voice quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now. I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as _7XXX and all users can then be registered with the server and get allocated the their individual numbers by the server.(till now i define the numbers with callerid field). i need to do this so that i can add 15+ users without having to add each individually. what would be the entry inthe i...

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called PJSIP/99/sip:99 at 192.168.1.73:5060 -- PJSIP/99-00000023 is rin...

...1] type=registration transport=upd_static outbound_auth=10001 server_uri=sip:600 at 192.168.1.1:5060 client_uri=sip:600 at 192.168.1.4:5060 [10001] type=auth auth_type=userpass password=600 username=600 [10001] type=aor contact=sip:192.168.1.4:5060 [10001] type=endpoint transport=upd_static context=dialmap disallow=all allow=ulaw outbound_auth=10001 aors=10001 [10001] type=identify endpoint=10001 match=192.168.1.1 when I call 600 from other pbx I getting an notice NOTICE[10202]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '"Ilya" <sip:502 at 192.168.1.1&...

...m ?99" and MeetMe ?1010" now I calling 99 -> 89264959635 via 10000 /pbx/agi.php: [agi_channel] => PJSIP/99-00000012 /pbx/agi.php: [agi_callerid] => 99 /pbx/agi.php: [agi_calleridname] => 99 /pbx/agi.php: [agi_context] => dialmap /pbx/agi.php: [agi_extension] => 89264959635 then I would like to direct transfer this call to 1010 and when I do that from my phone I getting this agi_request in AGI: /pbx/agi.php: [agi_channel] => PJSIP/10000-00000013 /pbx/agi.php: [agi_callerid] =&...

...g: [udp] type=transport protocol=udp bind=192.168.1.4 local_net=10.0.0.0/24 local_net=10.0.1.0/24 local_net=192.168.1.0/24 external_media_address=195.239.8.122 external_signaling_address=195.239.8.122 [udp_B] type=transport protocol=udp bind=192.168.53.9 [10000] type=endpoint aors=10000 context=dialmap disallow=all allow=alaw,ulaw transport=udp_B [10000] type=aor contact=sip:192.168.53.1:5060 max_contacts=4

On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <jleed at me.com> wrote: > NAT endpoint calling local endpount - switching to native_rtp then no audio, > both of them have direct_media=no, Verbose log: > > -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in > new stack > -- Launched AGI Script /pbx/agi.php > -- AGI Script Executing Application: (Dial) Options: > (PJSIP/99/sip:99 at 192.168.1.73:5060,20) > -- Called PJSIP/99/sip:99 at 192.168.1.73:5060 >...

...4 > local_net=10.0.1.0/24 > local_net=192.168.1.0/24 > > external_media_address=195.239.8.122 > external_signaling_address=195.239.8.122 > > [udp_B] > type=transport > protocol=udp > bind=192.168.53.9 > > [10000] > type=endpoint > aors=10000 > context=dialmap > disallow=all > allow=alaw,ulaw > transport=udp_B > > [10000] > type=aor > contact=sip:192.168.53.1:5060 > max_contacts=4 >

Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries