search for: yourcontext

Dear list, following problem, i have some users, who are supressing their callerid. This setting is adjusted at the sip phone. So if these guys are calling internal persons nobody sees the callerid. I am looking for the following resolution: User has set his phone to anonymous, user calls somebody internal, Asterisk initials a lookup on the channel and generates a new callerid for the

I have in my sip.conf the following [jon.moore] type=friend mailbox=8100,8150 In voicemail.conf, both mailboxes are defined. On my Aastra 480i phone, I only see the first mailbox listed. I've verified this, by changing mailbox= to reverse the order, and I then see 8150 when I go to Services > Voicemail on the phone. I also only get MWI events for whichever mailbox is listed

...party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and replace "yourcontext" for your dial plan. [blueface-in] type=peer host=sip.blueface.ie context=incomingpstn And then in my extensions.conf to have something similar to the following (or however I wanted to handle my incoming calls) [incomingpstn] exten => 2093,1,Wait(1) exten => 2093,n,Background(Mai...

Hi everybody, How can we add new contexts in asterisk realtime module? All I could figure out after googling is that a new context HAS to be declared in extensions.conf with 'switch => Realtime/@<databasetable>' under the context name declaration. This works fine as long as we are adding extensions only to this one context, but doesn't give the freedom to add new contexts for

...party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and replace "yourcontext" for your dial plan. [blueface-in] type=peer host=sip.blueface.ie context=incomingpstn And then in my extensions.conf to have something similar to the following (or however I wanted to handle my incoming calls) [incomingpstn] exten => 2093,1,Wait(1) exten => 2093,n,Background(Mai...

First of all great job on diax. I downloaded it and tried it, could not connect, got an authentication rejected,but I have not had a chance to figure out why yet - tried with a working gnophone setup in the configuration files. Is there any way to pass command line arguements to the program ? Where I see a real niche for a lightweight softphone is being able to serve the thing from a

Skipped content of type multipart/alternative-------------- next part -------------- _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

Hi Guys, Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and 6730i, but none of them indicate the voic-email. Where should I look for trouble to find the root issue for MWI? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100714/d268d1f8/attachment.htm

I need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15

I'm trying to connect my Asterisk server via sip using my vonage soft phone account. Has any anyone successfully got to work? I get error from asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: Unable to lookup '216.115.25.199:5061' when trying to register with

Hi, How do I disabled asterisk to use database and storage voicemail in directory. Im getting the below error [Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353 last_message_index: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04]

...ER and use Asterisk for PSTN access, voicemail etc >> >>My provider told me to change my sip.conf as follows >> >>register => username:password@sip.blueface.ie/2093 >> >>; To receive incoming calls specify this block and replace >>"yourcontext" for your dial plan. >>[blueface-in] >>type=peer >>host=sip.blueface.ie >>context=incomingpstn >> >>And then in my extensions.conf to have something similar to the >>following (or however I wanted to handle my incoming calls) >> >>[inco...

...5064 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm srvlookup=yes canreinvite=no; autocreatepeer=yes nat=yes dtmfmode=info insecure=very registerattempts=0 register => username:password@sip.blueface.ie/1234 ;To receive incoming calls specify this and replace "yourcontext-pstn" for your dial plan [blueface-in] type=peer host=sip.blueface.ie context=pstn [1234] type=friend username=1234 canreinvite=no context=pstn insecure=very ;callerid= "Ais" <1234> host=dynamic nat=yes dtmfmode=INFO mailbox=1234 disallow=all allow=alaw allo...

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro Some of the features are: - registering with Asterisk PBX; - can use any audio device as ring device (including PC speaker), independent of the play device;