similar to: reducing iax packet size

Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys | outkeys | auth | accountcode | amaflags | callerid | context | defaultip | host | language

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

I saw this thread which is very similar to my issue, though I cannot solve mine with iptables. http://lists.digium.com/pipermail/asterisk-users/2013-September/280429.html Using asterisk 11.5, IAX does not seem to be able to receive any packets. My IP tables looks like this: root at dlaptop:/home/darryl# iptables -L Chain INPUT (policy ACCEPT) target prot opt source

Package: logcheck Version: 1.3.15 Severity: minor Tags: patch Hi, Got this log from time to time in System Events: Oct 23 13:48:16 pig2 named[28880]: success resolving '26.0/25.218.183.203.in-addr.arpa/PTR' (in '0/25.218.183.203.in-addr.arpa'?) after reducing the advertised EDNS UDP packet size to 512 octets Changing the regexp for the "(in '...'?)"

Hello, I want to reduce the bandwidth taken by an IAX trunk when used with a small number of voice channels. When only one call is passed through an IAX trunk, the IP overhead is indecent. I would like to increase the IAX voice packet emission interval (20ms - i'm using speex) to something much larger, in order for the packets to transport as much data as possible. I've been

Hi all, I have spent some time searching, but I haven't found a way to prevent * from concatenating two frames into one IAX packet. I have a situation where I make an IAX GSM call to *, which transcodes to an iLBC SIP call. Every second voice packet the IAX client receives contains 2x 20ms frames, the other containing only one. I presume this is related to the mismatch of 20ms GSM vs

I have, in iax.conf the register statement: register => username:password@iaxtel.com This causes registration attempts to iaxtel.com for both IAX and IAX2. Every once in a while there is a packet for port 4569 keeping the IAX2 registration alive. This is fine. But, I have a barrage of registration attempts to iaxtel on port 5036 for IAX. Every UDP packet is answered with an ICMP packet

I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get iax to work. I've opened 4569 in the EC2 Security Group. I'm using the zoiper client. Using tcpdump I can see the zoiper packets coming in on 4569, but nothing shows on the asterisk cli. Frame 33: 79 bytes on wire (632 bits), 79 bytes captured (632 bits) on interface 0 0000 12 31 3b 12 40 84 fe ff ff ff

hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and ...... best MAni ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097

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

We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats" on client machine shows more and more dropped packets on the

Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.

I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for

Hi everyone, THis is my second thread regarding the issue.(before I was having problems with accessing my email, which slow down my responses, sorry for that) My setup looks like this Firewall | | Asterisk---Asterisk (two asterisk servers with the same setup for high avail) | | phones Ports 5060, 10000-20000, 4569, 5036 are forwared to 192.168.1.251 which is virtual ip address on one of the

I'm trying to set up * server behind NAT. The box is set up as DMZ in my DSL router, i.e. all incoming connections without explicit port mapping are forwarded to *. So far I'm unable to get this setup to work for either IAX or SIP (tried IAXComm & XLite softphones on public IP address). Data seems to come in fine (IAX/SIP debug shows message interaction taking place), but there is no

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Hi, We are use IAX protocol between two asterisk servers. Now we send information through this protocol by using EXTEN We see that the variable EXTEN only holds 66 characters. If we set a value larger then 66 characters, for example 70 characters. The last 4 characters are cut off. Is there a way to increase this variable? Kind regards -------------- next part

For those that might be using Cisco 7940/7960 sip phones and placing calls across an iax2 link, we think the voice quality problem has been identified and corrected. The dev cvs should be updated as of about 3:30pm CDT today (April 14). History: Calls originating from a Cisco 79x0 sip phone and sent via iax2 link to some distant * machine resulted in very poor quality audio, and in some cases,

Howdy! I'm still trying to make it onto the 1.4 releases. Almost ready to make the switch, but here's one last thing that doesn't seem to work: Server A calls Server B over IAX, i.e "Dial(IAX2/SrvB/${EXTEN},60)". Both machines are set up in iax.conf to use RSA certificate-based authentication. The public keys have been exchanged. This setup works just fine as long as Server