search for: nhadi

Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie

...sion 101 and 102, customer B will have a different trunk 87659043 but will also have the extension 101 and 102. i want to create a billing system to monitor only the trunks and also to load amounts on those trunks. is this possible? will i be able to use app_prepaid for this? thank you. regards, nhadie --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080422/e75810aa/attachment.ht...

...tbound i will make him use of [sip-trunk-100] another user, 101300 when that users calls outbound i will make him use of [sip-trunk-101] actually the 100 and 101 at the beginning of the username is the accountcode i used for cdr. hope my question was clear enough. thanks in advanced regards, nhadie

Hi, Anyone knows a DID provider that can do both outbound and inbound? Regards Nhadie

...nything that would not make my dialplan to wait for 30 secs. also i'm not using rtcachefriends, how would i know in the CLI which user is registered? i tried sip prune but it shows me nothing sip prune realtime peer all No peers found to prune. anyone experienced this? thank you regards. nhadie

...y this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing "bs523450017" instead of +6523450017. i tried putting it inside double quotes CALLERID(num)="+6523450017" telco says the same thing. is this possible? thank you Regards, nhadie

Hi, I'm using gotoiftime on asterisk, but it seems  there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Regards, nhadie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080611/63e105bd/attachment.htm

...erisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start troubleshooting this? TIA Regards nhadie

...still likely run into NAT issues that need to be dealt with somehow) and still let you use Asterisk as an in-between point. Together, Asterisk and SER make a very powerful combination for providing a full suite of services to clientele, and each plays well off the other's strengths. N. Nhadie wrote: > Hi All, > > What's the advantage of combining ser with asterisk? I always see > comments like using ser with asterisk is a very good solution etc. etc. > the thing i liked with ser is that it does not do codec translation, > which saves me cpu usage and also bandwi...

...nd route 1 extension that starts with 4 goes to outbound route 2. Basically, i'm hosting two(2) office, extension 3XXX is office 1 and extensions 4XX is office 2, they both have the same dialling pattern so i need to choose route based on source. i'm using freepbx for this. thank you Nhadie

...ove the timeout if there is any? thanks [Nov 25 13:22:37] ERROR[21026]: cdr_addon_mysql.c:171 mysql_log: cdr_mysql: Server has gone away. Attempting to reconnect. [Nov 25 14:20:32] ERROR[21061]: cdr_addon_mysql.c:171 mysql_log: cdr_mysql: Server has gone away. Attempting to reconnect. regards, nhadie

...7 extension 987654321 at outbound-route What i'd like to be able to is instead of a local extensions i would call an outside number then connect it another outside number. e.g. originate SIP/85431210 at outbound-route extension 987654321 at outboudn-route is this possible? thanks. regards, nhadie

...3] -- Called 103100 it stops there until it reached the timeout i set then it will say unavailable. is there a way that realtime will know that the phone is not registered anymore? or what could be causing the retransmitting of NAT? has anyone encountered the same prob? thank you regards, nhadie

Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing

...k to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080625/c0f6ea78/attachment.htm

Hello Nhadie, I had a very similar situation. My solution, even tough might not look very wise, solved my problem the way I needed. I repeated the GotoIftime command in the next line in my extensions.conf . Like this: GotoIfTime(22:00-23:59|*|30|jul?test,s,1) GotoIfTime(00:00-02:00|*|31|jul?test,s,1) Rgs...

...mething whereby a user can assign his extension on an IP phone in the office, and assign the same thing maybe to a softphone on his laptop or maybe a sip client on a mobile phone. so that whenever he leaves the office he can still be reach on his extension via the sotphone. thank you. regards, Nhadie

...ill have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i cant make calls. but if server 2 is inactive, asterisk does not run, as on the config it is binded on the secondary ip. anyone uses heartbeat for failover? tia. regards, nhadie

....204.205: bytes=32 time=43ms TTL=56 Reply from 202.203.204.205: bytes=32 time=12ms TTL=56 Reply from 202.203.204.205: bytes=32 time=13ms TTL=56 even if the latency is high i still have internet access as i can still browse and using yahoo messenger. anyone encountered something similar? regards, nhadie

I have a PRI connected to a traditional PBX using NI-2 and a typical config (further below). When I call from a SIP/IAX phone to an extension on the PBX, only the number makes it through. If I plug that same port on the PBX to a carrier the PBX presents both name and number. Hints or pokes to relevant chapters in documentation? My config is essentially like one found here: