similar to: Registrations problems to SIP-provider.

asterisk*CLI> sip show domains Our local SIP domains: Context Set by jocan.local (default) [Configured] 192.168.1. (default) [Configured] [Jun 26 17:49:03] NOTICE[5570]: chan_sip.c:15889 handle_request_register: Registration from '<sip:grandstream at 192.168.1.248>' failed

I have installed gnutls and gnutls-devel from RedHat repositories [root at asterisk asterisk]# yum install gnutls gnutls-devel I have installed iksemel with gnutls support : [root at asterisk asterisk]# cd /usr/src/iksemel-1.3/ [root at asterisk asterisk]# ./configure --with-gnutls --prefix=/usr [root at asterisk asterisk]# make [root at asterisk asterisk]# make check [root at asterisk

Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register => 092779077:XXXX at 85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret=XXXX fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes

On my Grandstream GXP 2010 I have the possibility for 6 channels and thus 6 different accounts... Line 1 I define an account that registers directly to an online Asterisk-server, somewhere in a datacentre. Line 2 I define an account that registers to the local Asterisk-server (NSLU2 unslung) When I activate both accounts, only the first account (to the Asterisk-server on the internet) registers.

Every morning I check my SIP registry to the SIP-provider. And I must conclude that during the night somewhere registry has failed. asterisk*CLI> sip show registry Host Username Refresh State Reg.Time 85.119.188.3:5060 092779077 105 Failed Sun, 05 Jul 2009 23:11:40 asterisk*CLI> sip reload [Jul 6 10:30:43]

Problem 1 : Incoming conversations from the SIP-provider come into the [default]-context and to the 's'-extension. I am unable to change this, even if I have : sip.conf [general] ;context=default ; Default context for incoming calls register => 092779077:XXXX at 85.119.188.3 ; incoming [092779077] type=user host=85.119.188.3 context=from3starsnet So I define no

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Hi list, I'm stuck for the moment @ the following : My Query (in a macro) : exten => s,n,MYSQL(Query resultid ${connid} SELECT\ vakantie_set\ vakantie_data1\ vakantie_data2\ FROM\ AstDB\ where\ SIPACCOUNT="${ARG1}") Asterisk CLI : [Aug 30 14:07:42] -- Executing [s at macro-vakantie:2] MYSQL("IAX2/zoiper-9238", "Query resultid 1 SELECT vakantie_set

Hey list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A --> Z. In my sip.conf : register => user1:passwd1 at server/user1 register => user2:passwd2 at server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer

I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered

Hi, I was trying to get an image/pdf of a sequence of colors in the order given. For example: myCols <- c('#0000BF','#0000BF','#0000FF','#0000FF','#0000BF','#0000FF') I'd like to make a strip of colors as they appear in the order above and save it as a pdf file. Is there a function in the base package (or some other package) to do this?

I would like to assign a variable y the string: <question><span style="color: #0000FF; font-weight: bold; font-size: 14pt"> How can I do it - is it possible? Best, Robert

Hello: I want to have a local telephone number that, when the people calls this number (via mobile or normal PSTN), the voip provider stablishes a SIP session to my asterisk box. It is possible? If yes... What providers have this service in Europe? It is difficult to configure and get things working ok? Will my asterisk box see the mobile or normal PSTN phone# that is calling the number

Is there any easy way to identify which script or service is connecting to the Asterisk manager? Somewhere on my system a script or service is trying to connect with a bad user name or password. I get the following error: connect attempt from '127.0.0.1' unable to authenticate I thought maybe I could do a tcpdump on port 5038 and try to fish out the bad username or password but I

Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software.

All, Having trouble getting answer from Verizon. I believe Asterisk will let me specify a name and number that is sent to the PSTN (Verizon) of outgoing calls. For instance, if I have a client, First Bank, and their toll free number is 888-555-1234, I could send that name and number. Verizon is telling me that they will forward the number I send them, but the name will be my company's

Hi, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>

hi, there: i have 2 questions associated with heatmap in heatmap.2{gplot}, there is a bar called "raw z-score" showing the coloring legend for each pixel. Where can I find that z-score's formulae? question 2, for example, I have 5 groups and I want to label each group with a color name from "#FF0000" to "#0000FF" evenly. so basically i need a vector like this: