similar to: SIP Registration

Are there are any xmpp developers on this list? I might have a small consulting project to build an XMPP chat application/(or even better alter off the shelf application with desired customizations) Email me for details. Regards, Dean Collins dean at cognation.net +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net

hi, for my multi-tenant pbx, i would like to approach prepaid like this: when a customer dials number, i have an AGI that will determine what country was dialed and retrieve the rate from the rate table, once the rate is retrieved, i will get the remaining balance of that customer nd compute how much time remaining based on the rte and the remaining balance. then i set that as an absolute

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, 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, 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, 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 All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is

What would it do if you exten => h,1,ResetCDR(w) exten => h,2,NoCDR() exten => h,3,DEADAGI(get-unqiueid.php) I have not tried it but in theory it should write the first CDR and then kill the write of the second NO ANSWER CDR. Let me know if it works for you as I may need to do it on some of my h exten code as well. Bryant ---------------------------------------- From:

hi all, i recently subscribe for an isdn and terminate it on a 3825 router. i used it as a sip trunk for my asterisk. i'm a newbie when it comes to ISDN. and i've been experiencing some issues: 1. Call Hangup: When hangup is initiated from the outside the extension (softphone/ip phone) does not hangup, is this normal? shouldn't asterisk hangup the extension as well when it

Hi! I use the following configuration to register my asterisk server to my SIP provider: register => 12345:passwd@sip.provider.com/12345 sip.conf: [sipout-test] type=peer username=12345 fromuser=12345 fromdomain=provider.com secret=passwd insecure=very host=sip.provider.com qualify=yes context=test-incoming extensions.conf: exten => 12345,1,Dial(SIP/10) exten =>

Dear all- I am have trouble in using the model="ht" option in function plm from the plm library. I am using Package: plm Version: 1.1-1; R version 2.8.1 (2008-12-22) running on a FC-8 linux machine. Here is what I am trying to do: ##---------------------------------------------------------------------------- R> ###Prob 6 Chapter 3 Use R! Applied Econometrics with R (Kleiber

Sorry for the empty message. Pressed the wrong button. I have been wrestling with a pretty generic Asterisk configuration (version 11.11.0 ) set up with FreePBX. The trunk SIP is setup to allow ulaw,alaw,gsm, Video is disabled. I was using Eyebeam and am now trying Jitsi. Jitsi has a number of codecs enabled - opus, SILK, G722, speex,PCMU, PCMA, iLBC, GSM, G723 and telephone-event The

Hey all We have been working with SIP for years. It has the potential to be better than Skype. It is really all in the implementation. Not all SIP soft clients are equal nor are the networks and computers they are running on. I will not bash Skype. We have tested it and in most cases choose not to use it. It has it's place and is good for the user that meets it's specific target

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

Hi, Is there a way to check on AMI if a user is currently engage on the phone? i would like to display on my portal whether a user is calling or not. thank you regards Ron

Hi, what software do i need to convert an mp3 to a g729 format? I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is

You might check your phones as well. We had this problem early on with a softphone and it was a setting in the phone that was set to hang up after 30 seconds of inactivity "in case of network disruption". For some reason it was detecting "network disruption" in every call even when the calls were proceeding normally. Unchecking this box solved the problem. It may not be

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.

can anybody identify why the CLI is issuing a failure message while debug shows everything is fine???? this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]:

Hi, I have installed asterisk real time and sip buddies information is being stored in DB. Now I have a question, Asterisk Realtime Server -A Third party SIP server-B Question: Is there any configuration in * RT that it can register with defined sip user on Server-B I was only able to find sip users information in DB not about user registration on other server. -ag -------------- next part