similar to: Question about codecs and interoperability with cisco AS5350

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

Help, I'm stuck. Lost in the woods. I have one Asterisk running on FreeBSD outside on the Wild Internet. One on the safe inside, behind a NAT firewall. The inside server registers with IAX to the outer one and can place calls. The outside one can't register to the one on the inside, since it can't be reached on the private network. Now to my problem: * How do I dial from outside to

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

Hi, I created an iaxtel account, and was given a password containing an "@" character. The directory pages imply that they change the web login password only. How do I reset my IAX password so that it is usable in the iax.conf file? Thanks, Steve

Hello, I am new to Asterisk, so RTFM answers welcome too (just include the FM's link :). I'd like to build a simple dialout server based on Asterisk. I installed 0.4.0 from package (a Debian SID machine, "server"). The client is gnophone (a Debian SID machine too, "client"). My modem is a GVC 56k voice modem connected to the server's serial port. I modified

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

My configuration is: [ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] -- [CISCO ip phone 12SP+/Skinny] When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN everything working ok (RTP is ok). But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone IP phone party can hear ISDN party, but ISDN (incoming) party canNOT hear IP phone party

Try sending 5350 config and oh323.conf, versions, etc... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj. Sent: Wednesday, December 22, 2004 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am currently interconnecting to a PRI using a Cisco AS5350. I'd like to be able to dial specific numbers out by a specific isdn channel, so for e.g. if I dial 999 01 12341234 it should send 12341234 out via isdn channel one from the Cisco AS5350. If somebody would be able to guide on this, it would be appreciated. Regards, Sahil Gupta VoiceValley

Hi, Just wondering if anybody has encountered a similar problem as I have with recieving calls on Asterisk from a CISCO AS5350 (over SIP). I have dtmf relay configured on the AS, however, when someone calls in from the PSTN sometimes their digits are inputted twice, which messes up the extensions. If there is a better way to terminate calls from a AS without using SIP, that would fix this

I've reading this lists few months. We are small company, that makes some system intregration, development and deployments in VoIP scene. Completely under linux. Today i have 6 machines with asterisk, huge test base - including devices like AS5350, Audiocodes gateways, ATAs, IP phones ... Now is time to make a decition for including * in our future projects. Main goal for us is the Stability.

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

Hi. I was wondering if there's a parameter to limit the number of concurrent sessions in IAX, globally or on a per-user basis. That could be needed for security purposes (to prevent dos attacks), to limit bandwidth / cpu usage, or to not allow more than N guest connections, for example. Any other VoIP channel support that? (like SIP, MGCP) Matteo. -- Brancaleoni Matteo

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hi all I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec. I need to get this down to 15 sec (nat /pat firewall issue) any ideas? thanks Liaan

Hi all, I'm using asterisk in a production environment now and this afternoon I got reports complaining that it was not working. Looking at the asterisk console output, I saw it contains lots of error messages as printed below. Unfortunately it is not obvious from the logs as to what started all this. Just before the error messages start, everything seems to be working fine with no problems.

G'day list, I am getting a lot[1] of traffic on my Internet link, ICMP messages from 69.73.19.178 telling me UDP port 5036 is unreachable (this IP address belongs to iaxtel.org). I see from the wiki that IAXtel supports only IAX2 from December 2003. Fine, however it looks like my * still wants to try and register using IAX1, and I can't find how to turn this off. This situation is