similar to: iax2 disconnect problem

The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though, the non-natted server is almost permanently _reachable_ from the natted server. Details are below

To answer my own message, I figured out a solution (untested) about 10 minutes after posting and leaving the office. Doh! Anyway, the solution (now tested) was to make the Asterisk server behind the NAT register with its peers. Despite reserving port 4569 in the firewall, that was not enough in this particular NAT firewall - it was only being reserved for one connection. Kind regards, Sebastian

Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been connected for few minutes) when an incoming call comes in. The log reads the following when this happens: [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963, having received INVAL [Nov 17 15:25:04]

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the

Hi, Does anyone know how to configure speex in asterisk 1.0.10? I've successfully installed it but cannot get any idea how to set the quality, etc.. Thanks Regards, Stevanus

Hi guys, I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at the third days I activated setting jitterbuffer=yes and suddenly there is no voice when the call is picked up. It's really weird as if asterisk stops sending rtp packet. I've checked asterisk log and found nothing suspicious. Just weird :S. I tried it in 3 asterisk server and all of them are having

Hi, Does anyone know how to make this thing (7940) work with asterisk (chan_sccp module) ? I've set the configuration according to the wiki and now the phone just keep asking for CTLSEP<xxx>.tlv from my tftp server. In the cisco's web interface, I found this in the device logs : 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 ...

Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : - when I called from 7910 to another sip phone in the same asterisk server, the call took place normally. - when I called from 7910 to another sip phone in different asterisk server, the call is answered but I cannot hear nor say anything. The phone just immediately lose its tone. - when I got

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

Apparently we've been having calls sporadically drop. We're using an IAX outbound trunk and SIP adapters on the inside. Below is a log excerpt detailing one of the calls which dropped, and it looks largely normal to me except for this: Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: IAX2/teliax-2 Jan 5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging

I've turned on debug in a (IAXComm based) soft phone. I see the phone sending pings to *. I see * getting the pings. For some reason, with iax2 show debug, I never see any response on the console from *. However, the phone shows a response with INVAL. Seems like an odd response to a ping request. I believed it should get an ACK. Is that wrong? If I should get an ACK, what could I have messed

I've been having problems with incoming IAX2 calls - some work, but a large fraction are answered with "dead air" or disconnects from my IAX provider. Disabling the jitterbuffer seems to eliminate the problem (so far)! Has anyone else seen this? I'm using 1.2.6, but I'm not sure what my provider is using. A snippet of the a failed incoming call IAX2 debug is attached

I'm going to try and lay out all the relevant information I have here in this one post. I can provide more info if necessary. ISSUE 1: Office A routinely looses connection to Office B. When typing IAX2 Show Peers, it will show as Unreachable. I issue IAX2 Reload and it will work again for 1-3 days (haven't narrowed the time down yet). My theory is that the DSL at Office2 is changing

I know in advance maybe I'm overlooking something stupid, however I'm really lost and cannot find the solution... Situation: - asterisk-1.2.13 on a linux box with no iptables active. - one iax2 peer defined - one wildiax phone running on my laptop the soft phone is configured to connect & register on asterisk, however, I cannot get it working. What am I missing? Please help!!

Hi people, i have problem with IAX2 between two asterisk PBX. When i try call some number i get "INVAL" packet, but when i try call same number via OpenVPN (is between this two asterisk) call is working fine.So i debug communications and here is my opinion ... Schema of connection: Asterisk1 -> ADSL router with NAT -> INTERNET -> Asterisk2 A)Calling directly via public

I'm trying to test IAX2 with FWD It registers fine but when I try to receive the call I get: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (38) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (39) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476

Hi, I'm running several asterisk servers in combination with dundi. The servers are in different data centers, but other than that they are running identical copies of the same os image, asterisk configuration, etc. One server acts as the trunk and is used to terminate pstn calls, and pass them on to another server via dundi, which then answers the call. I recently noticed that one of

[Sorry if this arrives more than once. I have sent this twice and it never arrived, despite other messages getting to the list O.K.] ----------- Hello, I would like an incoming caller to be able to choose from the menu options in my extension.conf below. Once They have dialled the appropriate digit, * should call two extensions simultaneously: one SIP phone on this * server, and one over a

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