similar to: Registration of multiple SIP-clients for the same extensions

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

Hello, I have a small LAN network connected through an Asterisk Server. When I try to make a call between two of the user pc's on this network I get a "401 Unauthorized" error. Would anyone know how to remove the Asterisk Authorization/Authentication? I am not sure if this can be done with an entry into the sip.conf file, or by other means. My sip.conf file is shown below: ;

Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch

hi i've got an AAstra 9133i ip phone, when i've bought it, i've set it to use a SIP/400 account on my asterisk, then, i've changed settings and i've set set phone to use a SIP/500 account . now, when i connect the phone to tthe network, it register itself on asterisk with both accounts!!! -- Registered SIP '500' at 192.168.100.188 port 5060 expires 120 --

" Hello, I have been trying to get my Grandstream GXP2000 phones to stop showing ringing state on monitored extensions. But no matter where I put notifyringing=no asterisk still sends the ringing state to the phones. Is this a bug I should report or is there another way around it. Here is how i have my subscriptions setup: extensions.conf [demo] exten => 6100,hint,SIP/100 exten =>

Hello, I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: "Oooh, format changed to 2". Would anyone know why

I am trying to set up the BLF on a GXP2000. Currently what I have is extensions.conf: [globals] polycom430=SIP/101 [internal] exten => 101,1,Macro(voicemail,${polycom430}) [macro-voicemail] exten => s,1,Dial(${ARG1},10,tT) exten => s,2,VoiceMail(u${MACRO_EXTEN}@default ) exten => s,102,VoiceMail(b${MACRO_EXTEN}@default) [ext-local-custom] exten => 101,hint,${polycom430}

Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This

Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI> show hints -= Registered Asterisk Dial Plan Hints =- 30@default : State:Unavailable Watchers 3 29@default :

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648 Peer User/ANR Call ID Seq (Tx/Rx) Format Hold

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting "Explicit MWI Subscription" to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, Lee

Hello Asterisk community, I'm trying to use BLF with Asterisk Realtime, i've been searching for some info but nothing seems to be clear, can anyone help me eith some ideas to make this work ok? I'va my dialplan with Realtime Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1

Can anyone help with BLF for Snom 300s ? (Asterisk 1.4.10.1) I've setup hints for a couple of Snom 300's but Asterisk doesn't send Extension Changed messages to subscribed phones unless the second 'line' button is used (I've tried Snom's version 6 and 7 and two difference 300s). On the Asterisk Console I don't see any message when picking up a Snom 300 and dialing

Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|