similar to: [Fwd: Asterisk client for ekiga.net NAT problem]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

Hi, I'm currently working on a setup between Asterisk and VoiceGenie (which is a IVR system). The way my setup is done, is that I have a PRI line coming in my Asterisk server, and then VoiceGenie is connected to Asterisk via SIP, like any other softphone basically. I'm able to receive calls in Asterisk and then link them with VoiceGenie. But one of my issues is that when I get an

Hi All, I've got a realtime queue in place (strategy is "wrandom"), and have added a member dynamically via "queue add member ". My agent shows in the queue, but when he gets the call is not recognized as "In Use". Here is the output from "queue show" prior to the call: *CLI> queue show QUEUE_3 has 0 calls (max unlimited) in 'wrandom'

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it

My internal lan is small, 100mb, all wired. aastra phones. sip show peers ....... 142/... 10.10.10.42 D A 5060 OK (136 ms) 144/... 10.10.10.44 D A 5060 OK (138 ms) 145/... 10.10.10.45 D A 5060 OK (133 ms) But pings are < 1ms: ping 10.10.10.42 ........ rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms Why are the sip latencies so

Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =>), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register => then how it will distiguish the IP address in the "host" at the [sip_trunk] is the IP address of the softswitch that need to

I have a 100MB internal lan. aastra's are wired. asterisk box is wired next to the switch. But look: sip show peers ........ 142/142 10.10.10.42 D A 5060 OK (137 ms) 144/144 10.10.10.44 D A 5060 OK (136 ms) 145/145 10.10.10.45 D A 5060 OK (168 ms) 150/150 10.10.10.50

I think I'm able to connect with Ekiga, at least it reports "registered". Curiously, when I exit Ekiga and switch to SFLphone, it isn't able to connect with the exact same parameters; it just says "trying" and never resolves. I'm not able to test outside connectivity because of too many hops: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:thufir

Hello, we have a problem with Dovecot 2.2.9 running on an AIX 7.1 and compiled with xlc. At first we configured passdb to use our ldap directory via pam and experienced an Internal login failure like the following one Jan 13 16:20:02 imap-login: Info: Internal login failure (pid=29818948 id=1) (internal failure, 1 successful auths): user=<user>, method=PLAIN, rip=xxx.xxx.xxx.xxx,

Trying to set up Junction Networks for outgoing on 1.6.1: extensions.conf: exten => _99X.,n,Dial(IAX2/jnctn_out/${called-num}) iax.conf [jnctn_out] type=peer host=iax.jnctn.net username= secret= qualify=yes I'm not using realtime. But CLI: -- Executing [99xxxyyyy at internal:3] Dial("DAHDI/1-1", "IAX2/jnctn_out/1wwwxxxzzzz") in new stack [2009-06-27

I need 100 DID's in a specific rate center (916-854-xxxx). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas

ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: <sip:test at ekiga.net>;tag=as64618445 To: <sip:test at

sip.conf.sample: ;register => 2345:password at sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. sip.conf: [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, all! I'm being based on this document [1] to send and to receive calls using ekiga.net. But I'm seeing, in an Asterisk console, several messages of this type: [Feb 17 21:19:15] NOTICE[11875]: chan_sip.c:7715 sip_reg_timeout: -- Registration for 'danib2 at ekiga.net' timed out, trying again (Attempt #4775) -- Got SIP

Hello list, I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use. I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP