search for: sipserv

Hi, Thanks for your response on the kernel switching.I was away and could not reply immediately. Right now, I am facing a differentissue. I have to set up DNS server using BIND on Centos 4.3. When Itype the hostname on Centos, it shows: sipserver.vodcalocal.com But the cli prompt has root at sipserver~ meaning only the sipserver part of the hostname is displayed. whyis this so? What is the actual hostname then? I see in the /etc/sysconfig/network that thehostname is sipserver.vodcalocal.com the /etc/hosts file has the followingcontent:...

...s strange as I did not find any mention of it anywhere in my search. Simply speaking, I have an external SIP proxy server which I am trying to configure for incoming and outgoing calls from my asterisk installation. So here is my configuration in sip.conf [general] register => user:secret:user@sipserver.com:8080<http://user:secret:user@sipserver.com:8080> as long as I have just the above entry, I am able to receive incoming calls. Now I would like to setup outgoing calls too. So I create a new section in sip.conf [sipserverout] type=peer secret=secret username=user fromuser=user fromdoma...

Hello, I?ve been using Asterisk 1.2.18 for a while, and today, with no apparent changes, I started receiving these messages: Aug 31 13:26:57 NOTICE[27528]: chan_sip.c:5495 sip_reg_timeout: -- Registration for 'user at sipserver' timed out, trying again (Attempt #19) All trunks and extensions went to: sipserver:5060 user 120 Request Sent 011 (Unspecified) D N 0 UNKNOWN Using ngrep I can see incoming messages to the server (port 5060), but no reponse...

...egister a sip user to sip server. I can see after registration * is sending periodic "SIP Request: OPTIONS" messages to server. but it's not getting back any response that should be SIP 200/OK as the documents say. 3130.299707 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip: sipserver.net 3131.299513 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip: sipserver.net 3131.299574 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip: sipserver.net 3141.300277 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip: sipserver.net 192.168.2.113 is LAN and 58.ab.cd.ef is...

Does anyone know where i can get DIDs in New Zealand. I am look for area code 06. -James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004

...when the latency was unusually high,my xlite's (i have 2 xlite) cannot register. but when the link suddenly went stable, the x-lite just registered. what i forgot to look at is if the registration packet is reaching my asterisks. ------ when xlite cannot register --------------- Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data: Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=651ms TTL=56 ------- when xlite can register ---------------- Pinging my.sipserver.com [202.203.20...

...<callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName>SIPSERVER</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool> <sipCallFeatures> <cnfJoinEnabled>true</cnfJoinEnabled> <callForwardURI>x--serviceuri-cfwdall</callForwardURI> <c...

...6 on Primary D-channel of span 1 > > > Script: > > #!/usr/bin/perl -w > > use Net::Ping; > use Socket; > > $polycompath = '/home/XXXX/'; # Where you keep your config files > $arp = '/sbin/arp'; # Location of arp command > $sipserver = '192.168.XXX.XXX'; # IP of asterisk server > > $phone = shift; > > checkphone("$phone"); > touch( arp2config("$phone") ); > > reboot_sip_phone( "$phone", "$sipserver", "Reboot" ); > > sub checkphone...

...o someone else can prettify it as they see fit. #!/usr/bin/perl # # PolyReboot.pl # # Reboots a Polycom 500 or 600 phone # use Net::Ping; use Socket; $polycompath = '/home/poly/'; # Where you keep your config files $arp = '/sbin/arp'; # Location of arp command $sipserver = '192.168.XXX.XXX'; # IP of asterisk server $phone = shift; checkphone("$phone"); touch( arp2config("$phone") ); reboot_sip_phone( "$phone", "$sipserver", "Reboot" ); sub checkphone { # Checks for existence of phone, makes sur...

Why is SER considered a better SIPserver than asterisk , why is it that SER can handle more clients than asterisk can. And if this is just cause of say poor SIP handling code in asterisk then is there anything being done to fix it. Just wanted to know why SER claims to be better than asterisk as a SIP server. ? -- regards Vikram (htt...

...ice phone from the asterisk PBX at all times. Now I have my Sip.conf setup with externip= X.Y.Z.250 [general] port = 5060 bindaddr = 0.0.0.0 context = default allowoverlap=no srvlookup = yes : externip = externip = x.y.z.250 localnet=10.202.17.0/255.255.255.0 qualify=yes nat=yes register = xxxxxxx:SipServer/xxxxxxxx limitonpeers=yes allowsubscribe=yes notifyringing=yes notifyhold=yes useclientcode=yes canreinvite=no I have pfsense setup to forward ports 5060 and RTP ports over UDP back to the internal asterisk server. And a firewall rule to allow this traffic from only my ITSP SipServer. I can mak...

...the install... I need recomendations in Linux distro... asterisk min. install ...etc..any info i can get. Thanks for any help Miklos Atenciosamente Cl?udio Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP ( BR - 55 11 3801-3702 ( USA - 1 360-968-1591 ( FWD - 64662 ( sip:ipfone@sipserver.com.br www.ipfone.com.br info@ipfone.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040616/33cd9fbd/attachment.htm

...************************/ <html> <head> <script type="text/javascript"> ? function webcall_win(sip,realm,phone,user,serviceName) { //You can have your ajax code here communicating with your site... //XMLHttpRequest... ? var URL? = "http://doddle.com.br/endoddle.jsp?sipserver="+sip+"&siprealm="+realm+"&callto="+phone+"&username="+user+"&provider="+serviceName; window.open(URL,"MyWindow") } </script> </head> <body> <h3>Your Asterisk?Applications web site...</h3> <p&g...

...EL, OPTIONS, BYE, REFER Contact: <sip:0170708959@192.168.1.4> Proxy-Authenticate: Digest realm="asterisk", nonce="2bc58039" Content-Length: 0 to 192.168.50.1:5060 Scheduling destruction of call '9663c234cb61f1a2@192.168.50.1' in 15000 ms Found user '4000' sipserver*CLI> Sip read: ACK sip:0170708959@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79 To: <sip:0170708959@192.168.1.4;user=phone>;tag=as6d84bb7a Contact: <sip:4...

...I get this error when trying to start asterisk: ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk What can be the problem? Thank you! Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 UK +44 870 - 3403539 FWD 64662 sip:ipfone@sipserver.com.br www.ipfone.com.br info@ipfone.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/72a54665/attachment-0001.htm

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

...ensions.conf ---------------------------- [default] exten => 1000,1,Dial(SIP/alex||t) [sip-in] exten => 1000,1,Dial(SIP/alex||t) [outgoing] exten => _0.,1,Dial(SIP/rix/${EXTEN}|20|t) ---------------------------- .qt/kphonerc ---------------------------- [Registration] AutoRegister=No SipServer= SipUri="Alex Polite" <sip:alex@localhost> UserName=alex qValue= ---------------------------- -- Alex Polite http://polite.se

...s to figure out which mailbox to enter. That's the basic idea. I have never used Asterisk before, and have (well had) no clue where or how to start, so I started in sip.conf, and just wanted to get the demo first. So I registered as ... register => mygenericforwardextension:password@legacy.sipserver.com:5060/9999 .. and then set up the 9999 extension in extensions.conf as shown ... exten => 9999,1,Goto(default,s,1) When I place a call to "mygenericforwardextension", the SIP signalling (INVITE) comes through to Asterisk just fine, but at the end I get a busy tone, and asteris...

...t a call reporting inbound calls are failing. When my server isn't registered, inbound callers get a "disconnected" message; very bad. When it happens, I've found that the IP address in the "sip show peers" output isn't correct; it doesn't match what "host sipserver" reports. Can someone explain to me what Asterisk does in the way of hostname-to-address lookups when it's registering with an external SIP server? What happens when the IP for the name changes? The provider claims they've changed nothing and a little cron job I setup to look u...