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...ve one, only a single Ethernet port and It is directly connected to my Linux box. My provider gave me a /24 subnet and 9 useable IP''s. # ip a s eth2 6: eth2: <BROADCAST,MULTICAST,UP> mtu 1500 qdisc pfifo_fast qlen 1000 link/ether 00:08:a1:72:c1:f5 brd ff:ff:ff:ff:ff:ff inet xxx.xxx.xxx.50/24 brd xxx.xxx.xxx.255 scope global eth2 inet xxx.xxx.xxx.51/24 brd xxx.xxx.xxx.255 scope global secondary eth2 inet xxx.xxx.xxx.52/24 brd xxx.xxx.xxx.255 scope global secondary eth2 inet xxx.xxx.xxx.53/24 brd xxx.xxx.xxx.255 scope global secondary eth2 inet xxx.xxx.xxx.5...

...lp will be appreciate Thanks Erick Weber VoIP*CLI> sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial("SIP/404-cbc9", "SIP/1088|60|tr") in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:1088@201.133.170.82 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: "Weber Automundo" <sip:404@XXX.XXX.XXX.XXX>;tag=as4da46cda...

Pardon the long log file but I'm fairly new to WINS servers and probably don't know what I'm doing. We've have two subnets (lets call them xxx.xxx.xxx.??? and yyy.yyy.yyy.??) I have one samba/linux server on both networks xxx.xxx.xxx.2 and yyy.yyy.yyy.2. xxx.xxx.xxx.2 is set up to be a domain master and yyy.yyy.yyy.2 is setup to be a local master with xxx.xxx.xxx.2 as its master. Things seemed to be working up until a couple of days ago...

Return-path: <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx> Envelope-to: xxxxx at xxxxxxxxx Delivery-date: xxx, xx xxx xxxx xx:xx:xx +xxxx Received: xxxx [xxx.x.x.x] (xxxx=xxxxxxxxx) xx xxxxxxxxx.xxxxxxxxxxxx.xx xxxx xxxxx (xxxx x.xx) (xxxxxxxx-xxxx <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxx...

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org

...ystem using auto-dialout files. I want the call to be terminated when a specific timeout (defined in the .call file) is detected. Both parties should then be hangup. The problem is that the timeout is never detected... How to solve this? Thank you, Pierre .call file ---------- Channel: IAX2/:@xxx.xxx.xxx.xxx/0111111111 Callerid: 111111111 MaxRetries: 5 RetryTime: 60 WaitTime: 30 Context: test Extension: 0222222222 Priority: 1 SetVar: ato=30 SetVar: act=testaccount extensions.conf --------------- [test] exten => _XXXXXXXXXX,1,SetAccount(${act}) exten => _XXXXXXXXXX,2,AbsoluteTimeout(...

...ket capture to see if there is a clue as to the cause of the > failure in the SIP signalling. Right, I see the following when running SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? <--- SIP read from TCP:xxx.xxx.xxx.xxx:50604 ---> REFER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/TCP xxx.xxx.xxx.xxx:50604;branch=z9hG4bK48c7492c From: "4005" <sip:4005 at xxx.xxx.xxx.xxx>;tag=203a07fceb4b00eff1377deb-da93e2ee To: <sip:4004 at xxx.xxx.xxx.xxx> Call-ID: OutOfDialog--001e-67a906f5-5333...

...successfully. Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK ---- SIP --- <--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:8005555555 at 64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555 at 64.2.142.93> Contact: <sip:15062fef-986e-4fcf-a93e-06b28da02fff at XXX.XXX.XXX.XXX:5060> C...

...ks, Aaron ======================================================================== sip.conf for user test [test] type=friend host=dynamic nat=yes canreinvite=no username=test secret=test ========================================================================== Failure REGISTER through Proxy: xxx.xxx.xxx.xxx = Asterisk yyy.yyy.yyy.yyy = Proxy zzz.zzz.zzz.zzz = User Agent Public IP 192.168.1.2 = User Agent Private IP <-- SIP read from yyy.yyy.yyy.yyy:5060: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKCefqJfo5hAO/paxgvw/iR7owic4= Via: SIP/2.0/UD...

...ver and a staging server and a production git branch and a staging git branch for the same repository on github. The production git branch and staging git branch are identical. I got my staging server running and during the second deploy, something unexpected started happening: ... ** [out :: xxx.xx.xx.xxx] Installing multi_json (1.5.0) ** [out :: xxx.xx.xx.xxx] ** [out :: xxx.xx.xx.xxx] Installing activesupport (3.2.5) ** [out :: xxx.xx.xx.xxx] ** [out :: xxx.xx.xx.xxx] Installing builder (3.0.4) ** [out :: xxx.xx.xx.xxx] ** [out :: xxx.xx.xx.xxx] Installing activemodel (3.2.5) *...

This may not strictly be an asterisk question, but not sure where else to post ... I have an Asterisk test server setup with two firefly clients, one on the local lan and one on an external ip address. Both clients are setup the same way and voice calls work fine. The asterisk console reports a "Registered" message for the external client at about one minute intervals but the

...caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the SIP Logs TestServer*CLI> <-- SIP read from 66.193.155.2:46478: REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0 Via: SIP/2.0/UDP 172.28.174.25:5060 From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX> To: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX> Call-ID: 74494a-1654e-43ce24ec@XXX.XXX.XX.XXX CSeq: 1 REGISTER Contact: "XXXXXX" <sip:XXXXXX@172.28.174.25:5060> User-Agent: XXXXX...

...one. Here is sip debug from pressing "transfer->blind->dial dest->Dial Key" (note both stations do have access tot eh dial-dst ext of 202010) <------------> -- Started music on hold, class 'default', on channel 'SIP/1050-0a6ffa70' <--- SIP read from XXX.XXX.232.66:8986 ---> ACK sip:1050 at XXX.XXX.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bK3dc3ce44DF7EE83D From: "1051" <sip:1051 at XXX.XXX.232.66:8986>;tag=D117C080-6FFBC539 To: "1050" <sip:1050 at XXX.XXX.232.175>;tag=as140f4415 CSeq: 1 AC...

...et prot opt in out source destination 0 0 DROP all -- * * 0.0.0.0/0 0.0.0.0/0 PKTTYPE = broadcast 0 0 DROP all -- * * 0.0.0.0/0 0.0.0.0/0 PKTTYPE = multicast 0 0 DROP all -- * * xxx.xxx.xxx.15 0.0.0.0/0 0 0 DROP all -- * * 192.168.9.255 0.0.0.0/0 0 0 DROP all -- * * 172.16.1.3 0.0.0.0/0 0 0 DROP all -- * * 255.255.255.255 0.0.0.0/0 0 0 DROP all...

...works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net SIP/2.0 Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK67c2df66;rport From: "asterisk" <sip:asterisk@82.XXX.XXX.XXX>;tag=as01265eaf To: <sip:freephonie.net> Contact: <sip:asterisk@82.XXX.XXX.XXX> Call-ID: 7263e88c20c9f38c34963cef6704cf07@82.XXX.XXX.XXX CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max...

Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. In our scenario the SP is sending call to our ser server and ser is forwarding the call to asterisk. In the asterisk debug I can see the DTMF keys are coming but ivr does not recognice those keys at all. I can see this in the debug. We are using ulaw and al...

...that is sent in response to a 200 message. The full SIP dialog is at http://pastie.org/private/nybdytnfyfenovpwfywcya so as to not clutter the email, but I have included the highlights below: >>> The call was ringing and is now answered: <--- Reliably Transmitting (NAT) to 82.158.83.xxx:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1226703311;received=82.158.83.xxx;rport=5062 From: "800902" <sip:800902 at 130.117.xxx.xxx;user=phone>;tag=467506068 To: <sip:6615xxxxx at 130.117.xxx.xxx;user=phone>;tag=as2e12c791 Call-ID: 21173886...

...elephone-event/8000 To: asterisk-users@lists.digium.com Message-ID: <465668BF.6080800@bingoconsulting.com> Content-Type: text/plain; charset="iso-8859-1" Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. In our scenario the SP is sending call to our ser server and ser is forwarding the call to asterisk. In the asterisk debug I can see the DTMF keys are coming but ivr does not recognice those keys at all. I can see this in the debug. We are using ulaw and al...

...o 7200 to the other end, but the Cisco 7200 does not return any audio back to the SIP Device or Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no luck. Has anyone else experienced this problem? My configuration and SIP debug is posted below. Asterisk server in SIP debug is xxx.xxx.xxx.xxx and Cisco 7200 is yyy.yyy.yyy.yyy. Thanks! IOS Config: Building configuration... Current configuration : 3362 bytes ! ! Last configuration change at 21:04:59 GMT Tue Nov 30 2004 ! version 12.2 service timestamps debug uptime service timestamps log uptime service password-encryption...

Hi, I have an eth0 interface (it's a CentOS 6 guest VM on a KVM host) which is configured as follows (see below) with a primary public IP address of xxx.xxx.xxx.130 (which works fine). I cannot set an alias IP address. I want eth0 to also use another IP address (xxx.xxx.xxx.131, so I create /etc/sysconfig/network-scripts/ifcfg-eth0:1. Then: # service network restart Shutting down interface eth0: Device state: 3 (disconnected)...