similar to: Dumb question... block 00

im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC

hi all, anybody have a siemens hipath 3500 with a sm2/pri card? because i need to connect to my box TE110P (e1) and i dont know how is the mode in the pbx to change it. thanks -- .- Pablo Allietti LACNIC

hi all i have an asterisk working and i need to add a mettme public service. for example i need to download a soft (sjphone) and without any configuration call to 509@asterisk.mydomain.com (meetme) and join a conference but when i do that i received an error saying nomber do not exist. but if i call a extension is work propperly. in the extensions.conf have exten => 411,1,Answer exten

Hi Everyone, I'm new to * and I'm trying to setup a small configuration of SIP clients. Eventually when I get this working I plan on expanding with a Digium developers kit to add analog phones and PSTN access. My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both peers seem to register with * but I cannot call to one another. When I dial the associated extension, the

Hi All, I was able to insert some extensions in Mysql DB and use them successfully. In Mysql extensions table the priority column is of type tinyint and when I give 's' value for it, it is not accepting that value as it takes only tinyints. Please tell how can I make that column accept values like t,s,i and make it work with asterisk in realtime without any problem? If I change the type

I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and fxoks configurations without avail. This is a single asterisk@home system with a single T1 card. Robbed Bit T1 ami, d4. ------------------inbound call

Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make calls on Audiocodes gateway . I got the folloowing messages on asterisk: [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80

On Mon, Apr 6, 2020 at 2:06 PM Administrator <admin at tootai.net> wrote: > Hello, > > We have a provider which is using Kamailio as front end. Our asterisk > 13/chan_sip server has no problem to register and pass/receive calls > form this provider. > > Now we want to move to asterisk 16/pjsip and face problem. Registration > is OK but when we pass a call our INVITE

Hello, We have a provider which is using Kamailio as front end. Our asterisk 13/chan_sip server has no problem to register and pass/receive calls form this provider. Now we want to move to asterisk 16/pjsip and face problem. Registration is OK but when we pass a call our INVITE never receive answer from the provider. We opened a ticket to their support but in the mean time we want to know

Our network is connected to a cablemodem using a dynamic DNS service to resolve our address. The Asterisk server has been alternately set up behind a NAT router and without a NAT router -- that is, with two NICs, one of which is providing NAT to the rest of the network; the office SIPs are behind that with static private IP addresses. Off-premise SIPs are all behind simple NAT routers.

Please help distribute the attached CFP Thank you Wael Dear Colleagues I would like to invite you to send a contribution to the IEEE 2006 Workshop on Signal Processing Systems (SiPS'06) Banff Park Lodge, Banff, AB, Canada October 2- 4, 2006 Please check the following website for more information www.ieee-sips.org <http://www.ieee-sips.org/> The IEEE

Hello, I'm having a look at section 13.1 from SIP Connect v2 doc (see [1]). It refers to RFC6442 which gives the following example (sorry for its length): INVITE sips:bob at biloxi.example.com SIP/2.0 Via: SIPS/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bK74bf9 Max-Forwards: 70 To: Bob <sips:bob at biloxi.example.com> From: Alice <sips:alice at

After reading this article: http://www.theregister.co.uk/2005/08/31/blocking_chinese_ip_addresses/ I got to thinking that there is really no reason for *any* traffic to hit my servers that comes from anywhere outside North America. So I wrote the perl script at the end of this posting to extract selected IP ranges posted at iana.org and convert them into iptables rules blocking any traffic

hi all i have a TE110P connected to my PBX when i try to call a extension number in other location 3525 the asterisk give me a error -- User entered '3525' -- Executing [450 at lacnicuy:4] GotoIf("Zap/31-1", "0?6:5") in new stack -- Goto (lacnicuy,450,5) -- Executing [450 at lacnicuy:5] Dial("Zap/31-1", "IAX2/lacnic:splacnic at

Rich, thank you for your informative reply. I checked with our admin and he replied: "I setup from the start "nat=yes" and "canreinvite=no" on sip phones from Internet and modified the rtp channels (voice ports) and the rtp port on the phones. Still have the same problem, no sound." Perhaps the VPN solution is something we should try but this is more limiting than

Hello all ! I have a problem with my IVR with terminate connection from PSTN phone Here is my configuration extension.conf [ivri] ;exten => s,1,Wait(1) exten => s,1,Answer ;exten => s,2,DigitTimeout(5) ;exten => s,3,ResponseTimeout(10) exten => ivr,1,Background(demo-congrats) exten => 1,1,MP3player,/mnt/linux/mp3/song/04.mp3 exten =>

On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Wow. Incredible. That worked. The backslash is important there; I kept > trying with no backslash and followed the instructions in > pjsip_wizard.conf.sample (in configs/samples) and it says we have to say > > transport=tcp ; the only example however talks about ipv4. > > Is

Hi All, I've got a really dumb question. I seem to have lost my Trash Can on my KDE desktop and try and as I might I can't get it back. Anyone have any idea how I can create this again or get it back? It's on the desktop as it should be in Gnome, but has disappeared from the KDE desktop. -- Mark "If you have found a very wise man, then you've found a man that at one time

Hello, Sorry for what might be a dumb question. I have a CentOS 4.4 machine with GD installed. I tend to prefer doing things via RPM to allow for updating in a easier fashion. I have a php based app that requires Freetype support in GD. This is a live server with many websites on it (although none are using GD at the moment. I am not sure the best/easiest way to get GD updated such that it

OK I have a very simple problem, hopefully somebody knows the solution. I have scgi Rails apps like this: app.domain.com and unaccelerated Rails apps like this: www.domain.com/app/ the problem only happens with the second type of app, and could have a lot to do with details with my host (a2hosting). www.domain.com/app/ works -- but www.domain.com/app triggers a "bad request" http