calling to sip uri 25.05.2006 09:10

Forum: Miscellaneous

I have to move abroad and taking out my laptop with asterisk pbx installed at it.
I have at office a DID number, for local city, a geographoc number.
Due of law regulations, i cannot use pbxes.com to access it, will not work, since foreign ip address.
I have cisco ata186 at office and all well working, just tested 1 minutes ago.
So, when i will take my asterisk out, people will be able still make and receive local calls at ata186, but!

i want to route sip calls from my account at pbxes.com there too.

since ata186 do not support registration at two gatekeepers at the same time - i can use only one feature

i can make calls to number@ciscoataip

it is ok for me if i will have only inbound calls

can someone explain me, is it possible to build some trunk, or something, to let call out sip uri, without registration at the pbxes.com

i see only one way right now, it is to use enum routing.

31.03.2006 23:11

Forum: Feature Requests

Zitat:

Originally posted by supernettel PayPal works for Me just fine, so I hope they do not abandon it!

PayPal do not work with all countries. So, I i will try paypal, it will be illegal

31.03.2006 23:09

Forum: Feature Requests

Agree totally. Already very good job. It is offers
Reg. A2billing - i have made it 'hosted' before. it is possible.

30.03.2006 21:35

Forum: Feature Requests

we hope that period will be extended and of course - direct debit from CC and www.moneybookers.com is only two methods available for me. + bank wire

30.03.2006 21:32

Forum: Feature Requests

guys can just install a2billing or any other billing and send automatically weekly CDR by email. yes, - life without CDR is not best

AMP message - nobody available to pick your call... 30.03.2006 21:30

Forum: Bugs

in my config i use call group; but anyway, sometimes call not processing by system (for example after reloading, sip peer not up and cannot send call to my phone). then i hear message from AMP: nobody available to pick your call...

it is not bug.

but very good idea is to use option noanswer in playback:

playback(no-body-available|noanswer)

then it will not cause phone answer on incoming call and will not charge caller.

wishes :) 27.03.2006 15:13

Forum: Feature Requests

great system. what additional rhings i expect in final version:

1. g729 and g723.1 (at least g729) codecs. maybe support for money. for them. just as example, provider from my home country giving me local phone and rates and only support g729 - www.ntt.lt;

2. jabber (with all modules- msn, yahoo, aol etc) + asterisk integration Jive software offer so nice server, it installed at my laptop and have some integration with asterisk.
this way userw will get more integration + possible to build own scripts (for admin of course) and do IM calling, IM callback etc;

3. billing. someone asked for a2billing. it really nice and i use it frequently. at least companies can be interested to cound costs - but to integrate to existing environment, a2billing need, of course, heavy patching;

4. some debug tool to watch related to mine accounts asterisk CLI output.

5. some modification of callback (now you use DISA, so easy to add small context and some ivr - will be clear for user) if need, i can send context.

6. i put it under #6, but i think it should be at beginning IAX2 support. Very important. Btw, only IAX2 provide good DTMF support, sip is not good with delays > 100 ms. your asian customers can be not happy.

7. ability to build own context and processing. of course, have to think about restrictions, and disallow some commands; or just created context have to be sent to admin/moderator for preview.

27.03.2006 09:59

Forum: Providers

only few operators allow to set own callerid number. first, because nowdays lot of routes going over E1 and even analog trunks and many providers just collect them at some softswith.
and only few do SS7 and have ability to set own callerid.

i kept sipgate at pbx, but it still not stable.

did number from sipgate.co.uk 27.03.2006 04:18

Forum: Providers

hi, wh else have UK DID's from sipgate?
i just register them at pbx, but it is not stable at all, always reloading ( i think ), i lucky when i can make inbound call

inbound route 27.03.2006 03:45

Forum: Bugs

i have number in UK.
Now i try to activate callback.
I made two inbound routes:

trunk name / calleris:

london / <empty> - routed to call group;
london / mymobile - routed to callback my mobile.

separately (if only one inbound route present) it works; if i put buth, always call routed to first (london/empty)

maybe it is not bug. but after few tests i do not found solution.

of course it is not big deal and i always can take DID just for callback, but

comment - after 10x tests it working, but so strange