similar to: Help with initial setup

Dear, When I start asterisk -vvvvvvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14)

Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

Hi all I have registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones. i configured the extensions.conf file in both the server. the extensions.conf file on server 192.168.0.9 is exten=>301,1,Dial(SIP/301@192.168.0.6,20,tr) exten=>401,1,Dial(SIP/phone1,20,tr) 301 is the extension number for phone 2 in asterisk server

Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast. My config is quite simple, I'm just following examples and the Wiki: I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to

I am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

hi. i have dialplan with 2 simultaneous calls - dial(sip/phone1&sip/phone2). when i cancel call on phone1 (push "reject" button), the call is still ringing on phone2 can i cancel call on both phones from one place(one phone)? thanks -- --------------------------------------- Marek Cervenka =======================================

Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to

Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very serious problem of modutils and iptable. Can anybody help me out. Thanx and Regards sanchal singh

Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. -Brian On Feb 11, 8:07am, "Juki" wrote: } Subject: [Asterisk-bsd] Asterisk not accepting multiple SIP phone logins } Hi all, } } I have Asterisk running on FreeBSD 4.x and I have made configurations to }

Dear All, I installed an Asterisk on a linux PC, and X-Lite on two Windows PCs, all in a LAN. But, when I make phone call from one X-Lite to another, I always get Call Failed: 404 not found. Here is my sip.conf: [Phone1] type=friend host=dynamic ;defaultip=192.168.1.103

I have an OpenBSD 3.7 gateway. This gateway run Asterisk. I have two windows box which use X-Lite softphone, and each box connect to Asterisk using this softphone (X-Lite). Asterisk use the following configuration : /etc/asterisk/sip.conf ; Phone #1 [Phone1] type=friend host=dynamic nat=yes defaultip = 192.168.10.12 # windows box IP context = sip callerid="Phone1" <1> ;

I have the following asterisk configuration (sip.conf) : [General] externip=82.79.81.3 localnet=192.168.10.0 localmask=255.255.255.0 [Phone1] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3]

Hi! I want to tell asterisk to simply pass-through any codecs that my phones support. I have to use codecs that are not popular and implemented by a third-party, asterisk has nothing to do with them. I've made a test with g722 (that asterisk doesn't support), i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it

Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

I don't know if this is possible, and I can't quite get my head around how to do it... If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: +---+ +---+ | A |-----| B | /+---+ +---+\ / \ Phone1 Phone2 Is there a way configure re-invites

Hello there, I appreciate any help about this problem that I can't figure out... I need to record all my calls: this is pretty easy using Monitor() before the Dial(). eg: exten => 425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb) exten => 425,n,Dial(${PHONE1},10) Now, I want to create a call group: I mean, I want a number (eg 800) that makes

Hello I have one phone (phone1) in one network, the other (phone2) in public network. both can call the other side; phone1 can be heard by phone2, phone2 can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER. Is NAT still necessary to be set on both phones? Thank you! steven

Hi, Actually on my setup all outgoing calls are going trhu a SIP unique account A have a second SIP account with another operator and I would like my setup to use alternatively each of the two accoutns Call 1=> Dial SIP/phone1 Call 2=> Dial SIP/phone2 Call 3=> Dial SIP/phone1 <...> If you have an sample please let me know