similar to: set callerid with plus sign

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:

Hi All, How can i play a sound during dial while waiting for it to connect? Coz currently i'm using SIP providers from other countries, when i send them the call there is a bit of delay to connect. I would like my users to hear a music first then when the call connects the sound gets canceled out. coz some users think the phone does not work coz they just hear a long silence but they

if i have this setup: [sip users] -- [asterisk] --- [as5300] --- [pstn] asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the asterisk so sip users can call out to pstn. what i would like to is do prepaid on those trunks, not on the sip users. sip users can call any other sip users . i want to do it that way coz i'm trying to build a multi-tenant pbx, and i will use

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

Hi, Is it possible to detect where the call came from and route it out to different sip trunks. e.g. i have user 100300 when that user calls outbound 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

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

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+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

Hi Friends, i have installed and configure asterisk-1.8.0. When i have tried asterisk start get below errors and not able to start asterisk. *FD 32767 exceeds the maximum size of ast_fdset!* Thanks in advance. -- Best Regards, Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, What does retransmitting NAT means? I have a client that uses SPA 942, and his phone sometimes cannot be called. i did a sip sebug and i keep on seeing retransmitting NAT. on the realtime it shows that it is registered, so when i try to call it , asterisk thinks it is still online so it tries to reach it instead of saying it's unavailable, [Oct 9 11:10:33] -- Called 103100 it

Hi, Need help on mysql cdr, i keep on seeing this log on the console. but my db is up and i see the calls being logged on the cdr table. is there a timeout when there is no activity? can i remove 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]:

Hi All, is it possible to choose outbound route by checking the extension of the caller? e.g extension that starts with 3 goes to outbound 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.

Hi All, I'm trying to use the orginate cmd. I have it working if originate is from a user e.g. SIP/7777 originate SIP/7777 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

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

Hi, I noticed that when dial terminates it does not return to the dialplan, and therefore can not execute any entry after Dial(). Is there any trick to overcome this limitation ? How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if I can not execute anything after Dial()? I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls end

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

i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part --------------

We have the possibly rather unique setup where we have cell phones posing as SIP devices. The SIP registration for those unfortunately doesn't go away just because the phone is off, since the registration is done by our cell-phone<=>SIP gateway, and that gateway has no way of knowing whether the phone is on or off. This is usually ok, but it gets problematic if the cell phone is a

Last week there was a thread on outlier detection. I came across an article which has a very interesting paragraph. The article is Missing Values, Outliers, Robust Statistics, & Non-parametric Methods by Shaun Burke, RHM Techology Ltd, High Wycombe, Buckinghamshire, UK. It was the fourth article in a series which appeared in Scientific Data Management in 1998 and 1998. The very

Hi, Is there a specific latency that asterisk accepts? I encountered a problem wherein when the latency was unusually high,my xlite's (i have 2 xlite) cannot register. but when the link suddenly went stable, the x-lite just registered. what i forgot to look at is if the registration packet is reaching my asterisks. ------ when xlite cannot register --------------- Pinging