similar to: IAX2 ping confusion and unreachable soft phones

I was wondering if anyone had seen this problem before and/or could offer any insight into what the trouble might be: I have an Asterisk machine that it set up as a mutual friend with another one (in another state... about 150ms away). Calls between the two fail to get established approximately 50% of the time. When a call works, everything is fine. When one fails, however, I see a large

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

Can anyone tell me why I keep getting these messages from IAXTEL? It does appear to register since I get lines like this: 2005-04-30 04:26:42 VERBOSE[1644]: -- Registered to '69.73.19.178', who sees us as 67.182.152.242:4569 But what is this? I don't think IAXTEL is working for me, since I can't dial 800 #s through it when I copy the iaxtel.com instructions. 2005-05-20

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

Hi, I'm using asterisk 1.2.7.1 and somehow my iax trunking is getting these problem :S. Sometimes iax acts weird and start to drop calls randomly and give these at the log: May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264]

I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived). I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server

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

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]

Some more info: Just tried this on a server without using any NAT and no port forwarding, no masquerading, and I still have the same problem. So there goes that idea. I do not know what this VNAK error means. By the way, I am using the latest version (1.2.6) of asterisk, have also tried other versions with the same problem [1.0.9 (Ubuntu Breezy) and 1.0.7 (Debian Sarge) and 1.2.1 (Ubuntu

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

Hello, I'm experiencing some problems with running IAX2 protocol on quite reliable link with G729A codec. My customer has 2mb FR link to the Internet used in about 20%. Ping statistics: 50 packets transmitted, 49 received, 2% packet loss, time 49496ms rtt min/avg/max/mdev = 9.308/13.126/33.307/4.851 ms Everything would be great, but the quality isn't good enough. I have 2mb/512kb DSL

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 run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse & repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 00000

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit

Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. This happens between 5% and 10% of the time. If they hang up and call

hallo, i am really confused how nativ briging is working with asterisk, i use a asterisk server as central server and register another asterisk and an iaxcomm client to the server, all three have public ips on the internet. somtimes, when i call from iaxcomm to my asterisk, the calls go peer to peer (i can see it with tcpdump) but sometimes the get routed through the central asterisk server

I'm having some trouble with placing some iax calls over an openvpn: Setup A is a 1.8GHz Celeron, T100P attached to a Zhone Zplex. Setup B is a 266MHz P2, T100P attached to a Zhone Zplex. Setup C is a 700MHz P3, T100P attached to an Adtran TA 750. Setup D is a 233MHz Pentium, with an X100P. Setups A and B are on the same physical network. IAX calls routed between them work fine. Setup D is

I had a working 1.0.9 asterisk installation and tried to get a Firefly IAX phone to register, but it was failing. I upgraded to asterisk 1.2.5 and the PBX is working fine, but the IAX phone still won't connect. Below is my iax.conf and the output from setting iax2 debug while the phone tries to connect. Could somebody please give me some pointers? This doesn't seem to be a normal

libswfdec/swfdec_text_field.c | 288 ---------------------------------- libswfdec/swfdec_text_field.h | 14 - libswfdec/swfdec_text_field_movie.c | 297 +++++++++++++++++++++++++++++++++++- 3 files changed, 291 insertions(+), 308 deletions(-) New commits: commit 2e79c953066b7f6857e88e7efaaed4d8846e9609 Author: Pekka Lampila <pekka.lampila at iki.fi> Date: Wed Oct 17