similar to: SIP -> H323 Seg fault.

we are trying to use asterisk for converting SIP to H.323 calls. asterisk (0.9.1) runs on the same linux (Redhat 8) box of our gatekeeper (gnugk version 2.0.8). the calls are going out through a cisco gateway. when I make a call from a SIP phone to a PSTN number reachable through the cisco gateway: asterisk diaplays Aug 5 23:24:26 WARNING[1255648560]: chan_oh323.c:2898

Using version 1.2.35 built by root @ slate on a i686 running Linux on 2009-09-15 00:24:10 UTC Problem - I cannot get featuremap right. Have added a feature that I want to direct to an extension in extension.conf Extension is 521 In features.conf - [applicationmap] dumpcaller => #9,callee,goto(521|1) show features - Dynamic Feature Default Current ---------------

Hi all, I really hope that you guys can help, because I've been tearing my hair out for the past 5 hours on this one. I have a Zaphfc (BRI) card in TE mode connected to the S-Bus of a Nortel Meridian phone system. Phone calls from the Nortel to say MSN 510 are correctly being sent to the right SIP phone. When asterisk dials say Zap/g2/224 (a Nortel internal extension) the call goes

Hello.. Maybe I'm asking something silly but..... How can I register my own app with * ? I've made a simple .so , but I cannot find it in asterisk when i type "show applications" Here is the code: #include <asterisk/lock.h> #include <asterisk/file.h> #include <asterisk/logger.h> #include <asterisk/channel.h> #include <asterisk/pbx.h> #include

Hello.. I have a linux box and I want to make priority on traffic generated by my LAN''s computers.. I don''t have a guaranted bandwidth, so I wanna use sfq... I want to make traffic to port 80 , 443 , 25 & 110 PRIORITY 1 Traffic src or dest 192.168.0.2 to make priority 2 And the rest to put it in proiority 3.. I did the following : tc qdisc add dev eth0 root handle 1:

== Parsing '/etc/asterisk/adsi.conf': Found -- Accepting call from '890003' to '185' on channel 27, span 1 -- Executing Answer("Zap/27-1", "") in new stack -- Executing Record("Zap/27-1", "soundexampless:mp3") in new stack -- Playing 'beep' WARNING[360468]: File translate.c, Line 128

Can I use a modem and a soundcard as an fxo ? I've read in the documentation something , but how can I do that ? Regards Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031107/2c4e43ce/attachment.htm

Hello.. I have a customer who wants to connect 2 PBX's over IP.. The setup should look like this: [PBX] <-- SS7 --> [Asterisk] <-- IAX --> [Asterisk] <-- SS7 --> [PBX] Since there are no SS7 cards , I was thinking at a way of carrying the E1 data as bulk...Can I do that ? How ? Is possible a scenario like this ? I'm thinking of IAX because I don't

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

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

Hello.. There is another way of doing SIP auth other then manually add the user & passwords to sip.conf ? I'm talking about radius or postgres.. Regards Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031026/3140157a/attachment.htm

Hello... Does anybody knows what is the meaning of thoose messages ? What i'm doing wrong here ? The * works fine , both incoming and outgoing services.. WARNING[163851]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 47 failed: Unknown error 500 PRI got event: 8 WARNING[163851]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 47 failed: Unknown error 500 PRI got event: 6

Hi, I install * and is working fine. I have 3 FXO cards w/ 3 phone lines. All the phones are SIP phones (Grandstream). The SIP phones from the same LAN w/ Asterisk are working but on the external phones (from the Internet) I don?t have sound. All the Grandstream phones from the Internet are register from different locations behind a NAT. All the sip users are register on * but the main issue is

Hello, I''m trying to build a replica volume, on two servers. The servers are: blade6 and blade7. (another blade1 in the peer, but with no volumes) The volume seems ok, but I cannot mount it from NFS. Here are some logs: [root@blade6 stor1]# df -h /dev/mapper/gluster_stor1 882G 200M 837G 1% /gluster/stor1 [root@blade7 stor1]# df -h /dev/mapper/gluster_fast

Hi I'm running Icecast2 with the IceS streamer which was installed from the ports on FreeBSD. Everything seems to be fine with my configuration as it starts up fine but as soon as I connect using a client like Media Player or WinAmp I see the following in the error.log [2003-10-27 21:00:27] DBUG format/format_generic_write_buf_to_client Client had recoverable error -1 [2003-10-27 21:00:27]

What is better? Cisco 7960 or Snom 200 ?? Bartosz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031028/befeaa4d/attachment.htm

We're being ping-flooded by the Nachi worm, which probes subnets for systems to attack by sending 92-byte ping packets. Unfortunately, IPFW doesn't seem to have the ability to filter packets by length. Assuming that I stick with IPFW, what's the best way to stem the tide? --Brett Glass

Hello Samba Team! I have a Samba fileserver installed on my network. Sometime I have a problem. There are some irresponsible users, who holds their network connections for no purpose. I want to disconnect forcedly such users . But I do not know how do it. Help me, please! Dementiy P. Saltaev, "Svyaztransneft", Volgograd, Russia, 01.27.2003; 13:41. -------------- next part

Hello Brian, Wednesday, October 29, 2003, 7:21:04 PM, you wrote: BRG> This anomaly is noted in the documentation for building the C front-end BRG> (http://llvm.cs.uiuc.edu/docs/CFEBuildInstrs.html); in step 4 it advises BRG> you to edit src/configure and "change the first line (starting w/ #!) to BRG> contain the correct full pathname of sh." I got the links to the proper

Hello.. How can I specify a class for htb based on a fwmark and user ip ? For instance: I have some routes marked with fwmark and their are very-high speed connections... But only to some IP''s.. For the rest , I must limit the user to 64Kbits Now , how can I limit the high speed connections ? I must create a rule and take in account both fwmark and IP ? To be more specific , I want