similar to: Help : problem in SLA (Shared Line Apperence

Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4? Specifically with the Aastra 55i. Specifically, I am using the Aastra 55i with the expansion module. We want to see if other handsets are being used. (BLF) Getting BLF to work would be a great start. It sounds like setting up the hints properly will achieve this. right? Not totally sure how this should be configured.

Hello Users, I Installed the Asterisk-1.2.11, For My Real time Use I'm Use MySql For Asterisk Database, By Using the Asterisk-addons -1.2.4 in My Linux. For My Voice messages Storage , I want To Use the MySql. In Googled it shows me the ODBC integration.. Is it need for that ODBC integration with MySql for my Voice Message storing in MySql. When I'm trying to integrate with ODBC +

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Hello Users, I googled Call Snooping, its shows only the it means, But i didn't find How to dialplan the Call Snooping, I seen that " What is Trixbox " in Asterisk I Use only some Feature in Asterisk (20), Is it need Asterisk to install the TrixBox in that same System where i installed the Asterisk Server Help me please :P -- Thanks and Regards Ravi Prakash Sunkara

Hello Users....... I'm Strucked in Call parking... I'm Using the Asterisk-1.1.11 version in My FC5 box, In That there is feature.conf I'm Using SIP channel By using Asterisk + OpenSER [general] parkext => 9006 ; What extension to dial to park parkpos => 9007-9009 ; What extensions to park calls on. These needs to be

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

Good Morning, I need a small help from U on Regarding the Asterisk , Currently I'm Doing Voice Mail in Asterisk which is forwarded By OpenSER. I can Leave the Voice message to the Caller , But Stores in this Directory " /var/spool/asterisk/voicemail/ " context. But For My Real Time and User interface developing , I want to Store in Database, As per My Knowledge and Googled ,

Fellow asterisk users, I am trying to get Single Line Appearance functionality working on a set of Linksys SPA942 phones and have not been successful. It looks like sla.conf is not getting read, only one phone reads as registered for the shared line, and a busy tone every time the shared extension is dialed. I have followed the documentation [1] and followed through other threads I saw

I tried to update and build 1.4 (SVN-branch-1.4-r45775) tonight. I took the additional step of nuking my modules directory first... When I used the command asterisk -v to start asterisk, it seemed to go along and get to the point where asterisk is running(ie Asterisk Ready). At that point it was eating all available CPU. I went ahead and tried to register a softphone to it via IAX2, which

Hi again, Kind of sheepish about asking for help, as I have only spent a day banging my head off this... I got my new Welltech 3701a, 1FXS,1FXO gateway. I flashed it with what is seemingly the appropriate firmware (SIP V1.04). This seems to have gone ok, and it is now registering both ports ok with asterisk. For 1 minute I thought I was home free and and everything was just going to work

Hi, Could you please help me!! I am trying to configure the Asterisk server. I have a analog phone connected to a FXS port of a Cisco 3745 router. This router is connected to a Asterisk server via Fast Ethernet interface. I am trying to make a call from the analog phone to a SIP client. This SIP client is registered to the Asterisk server. Analog phone number: 999 SIP client : 202 Sip client IP

Hello Users, Is Without Zaptel interface Installed, conference Bridge is worked or not. Why it need, For SIP conferences through OpenSER.... Please Help me For me its Giving Some Errors and warnings. *== Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 WARNING[12281]: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Oct 25

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

From eb20603913ff7350cd25b39d1eb37b8fddd16053 Mon Sep 17 00:00:00 2001 From: Zhang Xiantao <xiantao.zhang@intel.com> Date: Sat, 25 Aug 2012 04:11:08 +0800 Subject: [PATCH 2/2] Nested: VM_ENTRY_IA32E_MODE shouldn''t be in default1 class for IA32_VM_ENTRY_CTLS_MSR. If set to 1, L2 guest''s paging mode maybe mis-judged and mis-set. Signed-off-by: Zhang Xiantao

Hello Users, I doing on Voicemail in Asterisk For my RealTime, By using the ODBC connectivity For Voicemessages. in Made the Change in res_odbc.conf, odbc.ini, odbcinst.ini and voicemail.conf When I start My Asterisk server it give me Some Warning, When I googled , a proper Docummentation is not found, it found in some there languages, the First Warning is. Warning [30188]

Hello Users I'm Doing working on Both OpenSER and Asterisk ....... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=>115,1,Park() exten =>115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says " U parked on 701 extension " After When 9001 dial

Hello Users......... I'm Now doing on Asterisk Manager for My knowledge Growth, Can anybody explan me on Asterisk Manager settings....... in manager.conf [general] enabled =yes port = 5038 bindaddr = 192.168.2.75 displayconnects = yes [hyperion] secret = hyperion permit=192.168.2.76/255.255.255.0 deny=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user write =

Hello Users I'm planning to do Call Barging and Call snooping , I saw this Feature in asterisk.org. This Barging and Snooping are test for " is Agents are replying the Answer or not " that I'm guessing Can anybody help me... this Feature ...... How to do Call Barging and snooping in SIP Channels , I;m not using any Zaptel Card -- Thanks and Regards Ravi Prakash Sunkara

Hi, I have a strange error, that I haven't yet found in another posting on the archive. Here's my sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls callerid=No CallID [stefan] type=friend secret=blah username=stefan

/Hello Users, Good Morning, In Conferemcing How to Disconnect the phone while in between the Conference ..... When *I press the ' # ' key for Disconnecting the Conference.......... Below the Following to shows some Warning, ( in Red Color ) from-sip en *CLI> -- Executing Playback("SIP/9002-08f9feb8", "conf-hasentered") in new stack