similar to: why asterisk is replying 404 Not Found

hello any one using cvs version of asterisk(realtime addons). i have defined two users 2000 and 3000 in sip.conf. after that when i try to call 2000 from 3000 or try to call 3000 from 2000 it is giving me 404 Not Found error. Found user '2000' Looking for 3000 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP

I want smtp requests from the internet to address 202.1.2.3 are to be forwarded to 192.168.1.109, so I set ORIGINAL DEST is 202.1.2.3 but when I restart it show error: iptables v1.2.11: invalid TCP port/service `210.0.214.212'' specified Try `iptables -h'' or ''iptables --help'' for more information. ERROR: Command "/sbin/iptables -A net2loc -p tcp

hello i try to call from sip phone on asteris to open phone on GnuGK. can any one tell me why it is saying chan_oh323.c:2501 ast_oh323_new: Internal channel initialization failed. Bad binary? Mar 16 13:28:46 WARNING[5963]: chan_oh323.c:2727 oh323_request: Failed to create new H.323 private structure 4. Mar 16 13:28:46 NOTICE[5963]: app_dial.c:749 dial_exec: Unable to create channel of type

hello i want to rout my calls to h.323. i have registered my asterisk with GnuGatekeeper. but it is not routing my call to h.323 channel. he is saying Internal channel initialization failed. Bad binary? can any one check my settings what is problem here thanks in advance kamran exten=>_321XXXX,1,Dial(OH323/${EXTEN}@192.168.0.153:1719,30,r)

hi all i am using CVS with Realtime mysql on backend. Dial application is invoked again and again what is the reason. I have tested it by printing some message to debug. this application is invoked again and again here is debug you can see lot of messages from app_dial.c at the end. Any one tell me what is the reason. Is this a bug or what Kamran Ahmad

I simply cannot get X-Lite (Windows) or SJ (Linux) softphones (the only ones I have tried) to register with Asterisk on the LAN (no NAT, no routers). I have looked at every conceivable archived message regarding 401 Unauthorized, SJPhone, etc., and have looked at every relevant article in the Wiki (and then some), and it looks to me like everything should be fine....yet I cannot get these phones

Dear List members, After succesfully installing the * on a couple of systems, and putting them on test, I observed that there is an intermittent call drop on PSTN line. The systems are - Dell Optiplex P3/500MHz/128MB - Built-in ethernet - 1 X100P (Motorolla chip) card on PCI - 10G HDD etc. - Asterisk April 17 CVS. - 2 Mediatrix FXS ATA (4 phones) - 2 Grandstream phones. - sip.conf, zaptel.comnf

Greetings We just updated two RH9 servers to 3.0.14a-1 and are still seeing a lot of errors in the logs. They're not stopping us working but I would _really_ like to understand what is causing them. On the first server, in the logwatch, we get a whole swag of: samba: Unknown Entries: session opened for user janicer by (uid=0): 6 Time(s) session opened for user sonjap by

Hi, I currently have deadtime = 15 in my smb.conf. This featured worked good to disconnect clients that have been idle for > 15 minutes in the past. With the 3.0.13 version, it seems to do nothing. As in, it just keeps the user connected indefinitely. Here is what I see when no one is currently at the office and everyone is logged off their computer. Thanks [root@spicy p]#

Hi, all. I could not add any additional users to a file or directory using the Windows Security tab in Windows XP Sp2. Here is my configuration for samba-3.0.23c-2.el5.2.0.2. # grep ACL config-2.6.18-8.el5 CONFIG_EXT2_FS_POSIX_ACL=y CONFIG_EXT3_FS_POSIX_ACL=y CONFIG_FS_POSIX_ACL=y CONFIG_NFS_V3_ACL=y CONFIG_NFSD_V3_ACL=y CONFIG_NFS_ACL_SUPPORT=m # smbd -b | grep ACL HAVE_SYS_ACL_H

Hi All I have installed Samba 3 under RHEL v4, with LDAP support for the authentication and as a PDC Domain, it seems everything works fine, but the only problem was the network browser list. We have a different subnet 192.168.X., 192.168.Y., 192.168.Z., etc..., Samba local browser works fine, but the remote subnet and workgroup are inaccessible, I double check the nmbd logs, there an error which

Greetings, I'm trying to continue to do some processing after a TIMEOUT (absolute). In my dialplan below, when a call comes in to [default], I call macro-phonenum and pass it a timeout of 20 seconds. macro- phonenum sets TIMEOUT(absolute), then loops saying the phone number that was called (in MACRO_EXTEN). When the timeout expires I want to call my macro-hangup (so it can say

Mandrake 9.1 Samba 3.0, windows2000 pro sp4. I finally got Samba 3.0 working thanks to G?mes G?za. Now I am having another issue as outlined below: 1)There are 2 users on the Linux pc, markw (regular user) and root. Both have valid samba passwords. 2)I have a share called: [data] path = /test valid users = markw, root read only = No browseable = yes 3)I am able to map the share with the user

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

Hello, I'm trying to set up Samba + OpenLDAP as a PDC. I followed the instructions from chapter 6 in Samba-3 by Example, my system is SuSE 9.1. ldap, winbind, nmb and smb are running. testparm says my smb.conf file is OK. I set LDAP password using smbpasswd -w. There was a similar post a few days ago (smbldap-tools don't create machine account properly), but it didn't help me.

Friends-- I'm stumped! Cannot get mount.cifs to work over a tun connection. How would you trouble-shoot this? 1. It cannot be openvpn causing the problem: I can ping across the connection both directions on all machines. 2. It cannot be samba causing the problem: I can mount.cifs the smb shares on the lan (using the identical credentials file) without any issue. 3. Have iptables

Hi, I'm testing Asterisk with a new provider. On calls to US toll-free numbers, there is no audio (calls to normal numbers are ok). In response to a valid INVITE from Asterisk, something like this is received: SIP/2.0 183 Session Progress v:SIP/2.0/UDP [my public IP]:5060;branch=z9hG4bK62d91cea CSeq:103 INVITE i:7a1791cf52d6f3dc2d12b208051d0a21@[provider].com f:"Test User"

Greetings *`s, I have created a call file and am manually placing it in /var/spool/asterisk/outgoing, but I get the following errors in the log file : =========== Mar 16 15:26:57 DEBUG[2054]: Auto destroying call 'E36A0D74-1DD1-11B2-AC8B-860A22D62D71@192.168.204.10' Mar 16 15:27:43 WARNING[2054]: Unable to open /var/spool/asterisk/outgoing/chris.call: Permission denied, deleting Mar 16

Attached patch addresses following problems with pxelinux: 1. If a x0y number is present in ip address to be ipappend-ed, middle zero is omitted, e.g. 192.168.0.109 is passed as 192.168.0.19. 2. UDP port numbers allocated by pxelinux.0 are 100% predictable, which might cause trouble when client reboots too fast for orphaned tftpd process to time out. The latter problem is [suggested to

Set busydetect=no in your zapata.conf file. That should stop the random hang-ups. If you really need busy detection, try setting busycount=8 or even 10. If you still get random hang-ups, turn off busy detection and turn on call progress. May help the situation. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: