similar to: Park Craches asterisk

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! Maybe you can help me. 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, 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 possible that

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

I'm setting up a queue for two independent operator phones that are capable of answering multiple calls at once. It's currently working with the following settings and Asterisk 1.4: queues.conf: [telefonistas] strategy=roundrobin ;strategy=leastrecent music=default timeout=60 retry=0 maxlen=0 wrapuptime=0 ringinuse=yes autofill=yes joinempty=yes member => SIP/8899 member =>

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)

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

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've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband,

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

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]

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 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, I am using to Xlite to save video voice mail. when i retreive it, then only video show , no voice is here out. Plz tell me where ,i am wrong , and how i can able to see video plus here audio in voice mail box. I did following configuration In Sip.conf videosupport=yes [phone1] type=friend host=dynamic context= employees mailbox=101 at default

I am thinking that I have found a bug in Rails migrations. My app is using UUIDtools to generate guids for primary keys. To do this I pass :id=>false and then create my own id column as shown below. Next I leverage "execute" to create an index. It seems to work fine. The table in MySql is perfect. However the ID column and primary key on the ID column are not in the schema.rb file I

Hi! I am having trouble getting the following configuration to work: PHONE1 <-- rtp --> Asterisk <--IAX--> Asterisk_SRTP_1 <--- srtp ---> Asterisk_SRTP_2 <-- rtp--> PHONE2 This means, I am using regular voip clients without srtp support on both sides, but the communication between the 2 Asterisk_SRTP boxes must be secure. The Asterisk_SRTP_2 box is registered in the

Dear all, I've been taking my baby-steps toward setting up an Asterisk phone system in my office, as also between my home and office (connected by DSL). I'm have a rough time getting two * boxes talk IAX over a LAN. I don't know what I am doing wrong, but am attaching my iax.conf and extensions.conf on both the boxes. Does anyone see it? ------config files start------ site-0

Now that I am using the chan_sccp module, the phones now work as single line phones. However, these phones have support for multiple lines. So I setup phone1 with extension 1001, and phone2 with exts 1002 and 1003. If I call ext 1003 from 1001, phone2 rings correctly and if I pickup the handset on phone2 I can carry on the conversation. If I call ext 1002 from 1001, phone2 rings as it should

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 =======================================

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with

Hey, I have installed Ubuntu a few days ago, since my experience with Windows Vista has been horrible. I decided to run steam using wine, and I was also going to run Counter-Strike. So I installed wine first using the following command, "sudo apt-get install wine". Then downloaded steam and installed it. When I launched Steam, my friend's names were very hard to read. It's like