similar to: Ignoring too old packet packet

I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323

Hello, I have a problem with the following setup, I hope you can help me. I have two internet gateways, one for LAN1 and the second for LAN2. +--------------+ GW1 more eth0| |eth4(SNAT) GW2 ---...routers...-----+ router +----------------- | | +---+------+---+ eth1|

I used the "FreePBX on Debian" HowTo at http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles to initiate calls to my SIP carrier. They get my registration, but they see that my call is interrupted before they can complete the connection. My Asterisk log shows that the call times out after the time (45s) specified in my dialplan Dial() command. What is wrong? [from

I have a situation when I dial out my Zaptel I am getting a recording that I need to add a 1 or a 0 and the area code with this number. I have tried appending this and the number going out the zap is 1NXXNXXXXXX so it is going out with 1 and the area code. Someone has suggested that maybe the zaptel is dialing too fast. My question is how can I add a pause before dialing to test this out. I am

Hi, I need help finding why my system is dropping calls.. I enabled debugging on my box in the hope it would lead me to the answer as to why my system is dropping calls but unfortunately nothing is jumping out at me.. I have attached the portion of the messages file for two calls that were dropped.. (numbers names and ip's have been changes to protect the innocent) Can someone give me a

Hello, I am getting a bizzare amount of error messages in the log files. The system seems to be running fine...no one is reporting any issues and all calls are coming and going. System is showing higher than average memory usage. eth0 is showing a high number of errors Running v1.1 Has happened on older versions and have been seeing this for quite some time but have just now asked if anyone

I have found a very strange problem. We found that the time of establishing the websocket connection between mobile phone and server was too long. Then I use tcpdump to capture the data and found that the problem maybe has something to do with window scale option in SYN packet. Here is the SYN packet for websocket connection: 55488 ? 443 [SYN] Seq=0 Win=65535 Len=0 MSS=1460 WS=64

Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '765e20595817e9897b77cff23f821cc5@10.0.0.254' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '4858cde16223cc0716e325921a8a0654@10.0.0.254' of Request

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 have recently installed Asterisk@home and outbound calling is working great. However I am strugglings with inbound calls. I have setup a trunk for my provider, voipfone and in the inbound area on AMP I have the following :- user context name = 3011XXXX context=from-pstn dtmfmode=rfc2283 fromdomain=voipfone.co.uk host=voipfone.co.uk insecure=very secret=XXXXXX type=user user=3011XXXX

Hi, I'm using app_pppd with a Digium-PRI-card for PPP connections. I had some strange problems with some IP packets passing and some not, e.g. ftp login went well, but as soon as I tried to up- or download a file, noting was transferred. I finally guessed, it must have to do something with the packet size. Then I started pppd with the parameters mtu 296 and mru 296 as in further times with

hello everyone, Last week I installed asterisk 1.2.24 with digium TE220B card. I have a problem with our PRI and Asterisk: the call be interrupted.It happens either PSTN-to-SIP or SIP-to-SIP,almost every call. After spending several days searching on internet, I found a lot of discussion about this issue and I have tried many, but it still.I am totally new to Asterisk environment and suspect I

Hi list! Today I tried to change the NAT-configuration on my Firewall to use another port for SIP. I configured it so: /sbin/iptables -t nat -A PREROUTING -p udp -m udp --dport 10000:10100 -j DNAT --to-destination 192.168.20.120 /sbin/iptables -t nat -A PREROUTING -p udp -m udp --dport <my new port> -j DNAT --to-destination 192.168.20.120:5060 then, I tried to log on my Asterisk

Happy new year! The last 3 months my asterisk has run perferct. But after I have set 15 new SNOM 190 phones on it dies every hour. Nothing to se in CLI ore in the log. It dies with exit status 139 Is there anyone who has an idea of what is wrong - ore any tip on how to test. /var/log/asterisk/messages .................... Jan 1 15:18:17 DEBUG[7193]: Stopping retransmission on

Hi Folks, I'm trying to use my mobile as a trunk via bluetooth - calls done in a softphone go thru GSM network and calls destinated to my mobile are answered at the softphone. I have asterisk configured to do so but I'm facing an issue - Audio is audible but it?s not intelligible. I feel like the audio is breaking. Below is the asterisk log. I also get lots of ?hci_scodata_packet: hci0

Hi All, Was troubleshooting some odd VM network issues and discovered that we're seeing dropped packets + retransmissions across multiple domU OS's and dom0 hardware platforms. xendev01 ~ # tshark -R "tcp.analysis.retransmission " -i vif7.0 Running as user "root" and group "root". This could be dangerous. Capturing on vif7.0 3.054257

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

Well, I'm writing yet another queue_log analyser program in PHP, and I have noticed the following entry in my queue_log file from today: 1104796626|1104796618.532|queue|NONE|ENTERQUEUE||no 1104796664|1104796618.532|queue|NONE|EXITWITHTIMEOUT|1 So, pretty sure that I didn't make someone wait 30 minutes in my queue. extensions.conf snippet: [remote-oldnum] exten => s,1,Answer exten =>

This is a single aastra 9113i sip phone. asterisk 1.0.9 Why do I keep seeing this in the logs ? ------------------------------------------------------ Sep ?8 18:44:25 VERBOSE[18779]: Scheduling destruction of call '9157b7d5ef36e8ec556a68e446d4ad59@192.168.1.100' in 15000 ms Sep ?8 18:44:31 DEBUG[18779]: Setting NAT on RTP to 0 Sep ?8 18:44:31 VERBOSE[18779]: 11 headers, 2 lines Sep ?8

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