Hi, I have our FS server configured and running. I can register a polycom phone to it perfectly. Can make calls out and receive calls in. Works well, quality is solid, etc. Now, I want to register an asterisk box with FS. (Our eventual plan is to resell VOIP as a "trunk" to end users, so this is an important test.) FS won't route the calls from asterisk. I don't see any specific errors in the FS console, but the calls just die. Below is the result of "sofia status profile default" showing that both the asterisk box and polycom phone did successfully register. (username, password, host changed for privacy) Call-ID 456fa3406d67ee337c6c81264932f76a@127.0.0.1 User 3235551212@111.111.111.111 Contact "user" <sip: 3235551212@222.222.222.222:1024;fs_nat=yes> Agent Asterisk PBX Status Registered(UDP-NAT)(unknown) EXP(2008-09-22 12:17:40) Call-ID 8c5f3b02-c1f07a54-7d8ac0c7@10.0.1.110 User 3235551212@111.111.111.111 Contact "user" <sip: 3235551212@222.222.222.222:5060;fs_nat=yes> Agent PolycomSoundPointIP-SPIP_500-UA/2.1.3.0028 Status Registered(UDP-NAT)(unknown) EXP(2008-09-22 13:57:15) Below is the config in my sip.conf for asterisk. (IP and DID changed for privacy) [Freeswitch] host=111.111.111.111 username=3235551212 secret=password port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw Can anyone help me figure out what is wrong? Thanks, -N
you'll need to set from-domain in the sip.conf on asterisk ;) /b On Sep 22, 2008, at 2:30 PM, Noah Silverman wrote:
Below is the config in my sip.conf for asterisk. (IP and DID changed for privacy)
[Freeswitch] host=111.111.111.111 username=3235551212 secret=password port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw
Tried that and still doesn't work. I've attached the SIP INVITE so that maybe you'll see something that gives you a clue. Also, I don't know if it matters, but the FS server is actually in an off site data center. I'm connecting to it remotely from my office. Works fine for a single polycom phone. (That, and the sound quality is AMAZING! ) Here is what I have in Asterisk now... [Freeswitch] host=111.111.111.111 username=3235551212 secret=password fromdomain=111.111.111.111 port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw Here's the SIP INVITE. (IP's changed to protect the innocent.) 111.111.111.111 is the address of my FS server 222.222.222.222 is the address of my asterisk server 3235551212 is my username/did/account in FS U 222.222.222.222:1024 -> 111.111.111.111:5060 INVITE sip:13237773456@111.111.111.111 SIP/2.0. Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK03cd6fd2;rport. From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as146a87d7. To: <sip:13237773456@111.111.111.111>. Contact: <sip:3235551212@222.222.222.222>. Call-ID: 4382446d3a46269d7f469e93147b46eb@111.111.111.111 CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Remote-Party-ID: "3235551212" <sip: 3235551212@111.111.111.111>;privacy=off;screen=no. Proxy-Authorization: Digest username="3235551212", realm="111.111.111.111", algorithm=MD5, uri="sip:13237773456@111.111.111.111 ", nonce="c52ba984-f888-dd11-80f4-00188b37805b", response="01635afacb7eeebc6fc0888991c0411d", qop=auth, cnonce="4e065f27", nc=00000001. Date: Mon, 22 Sep 2008 22:54:14 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 234. On Sep 22, 2008, at 2:35 PM, Brian West wrote:
you'll need to set from-domain in the sip.conf on asterisk ;)
/b
On Sep 22, 2008, at 2:30 PM, Noah Silverman wrote:
Below is the config in my sip.conf for asterisk. (IP and DID changed for privacy)
[Freeswitch] host=111.111.111.111 username=3235551212 secret=password port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Is your asterisk server behind the firewall or NAT. Does your FS respond to the invite from asterisk, Did you run sip trace on both asterisk and FS. I mean it will be useful to know if the below sip trace is from asterisk or FS. If from asterisk, then need to make sure if FS got the request and how it replied to that request. Hope this will help you in debugging the problem. Jai www.didforsale.com *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com" On Mon, Sep 22, 2008 at 4:01 PM, Noah Silverman <noah@allresearch.com>wrote:
Tried that and still doesn't work.
I've attached the SIP INVITE so that maybe you'll see something that gives you a clue.
Also, I don't know if it matters, but the FS server is actually in an off site data center. I'm connecting to it remotely from my office. Works fine for a single polycom phone. (That, and the sound quality is AMAZING! )
Here is what I have in Asterisk now... [Freeswitch] host=111.111.111.111 username=3235551212 secret=password fromdomain=111.111.111.111 port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw
Here's the SIP INVITE. (IP's changed to protect the innocent.) 111.111.111.111 is the address of my FS server 222.222.222.222 is the address of my asterisk server 3235551212 is my username/did/account in FS
U 222.222.222.222:1024 -> 111.111.111.111:5060 INVITE sip:13237773456@111.111.111.111 <sip%3A13237773456@111.111.111.111>SIP/2.0. Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK03cd6fd2;rport. From: "3235551212" <sip:3235551212@111.111.111.111<sip%3A3235551212@111.111.111.111>
;tag=as146a87d7. To: <sip:13237773456@111.111.111.111 <sip%3A13237773456@111.111.111.111>>. Contact: <sip:3235551212@222.222.222.222<sip%3A3235551212@222.222.222.222> . Call-ID: 4382446d3a46269d7f469e93147b46eb@111.111.111.111 CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Remote-Party-ID: "3235551212" <sip: 3235551212@111.111.111.111>;privacy=off;screen=no. Proxy-Authorization: Digest username="3235551212", realm="111.111.111.111", algorithm=MD5, uri=" sip:13237773456@111.111.111.111 <sip%3A13237773456@111.111.111.111> ", nonce="c52ba984-f888-dd11-80f4-00188b37805b", response="01635afacb7eeebc6fc0888991c0411d", qop=auth, cnonce="4e065f27", nc=00000001. Date: Mon, 22 Sep 2008 22:54:14 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 234.
On Sep 22, 2008, at 2:35 PM, Brian West wrote:
you'll need to set from-domain in the sip.conf on asterisk ;)
/b
On Sep 22, 2008, at 2:30 PM, Noah Silverman wrote:
Below is the config in my sip.conf for asterisk. (IP and DID changed for privacy)
[Freeswitch] host=111.111.111.111 username=3235551212 secret=password port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw
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That SIP request was what was received on the FS server. I just made two calls. One from the polylcom phone which worked and one through asterisk which failed. I'm comparing the two SIP INVITEs to see what the difference is. (Looking at the Invites that FS RECEIVED.) Much is the same. here is what I found different. To: On the successful call has "user=phone" appended. Nothing on the failed call Call-ID: On the failed call is "abcdefetc:domain" On the successful call is "abcdetc:phone_ip" <--- Might be significant. Remote-Party-ID: Exists for failed call. Does not exist for successful call Proxy-Authorization: The uri for successful call has port 5060. The uri for failed call has no port <--- Might be significant. Still digging through debug stuff but wanted to send this to the list incase someone had any ideas. Thanks!! -Noah On Sep 22, 2008, at 4:36 PM, Jai Rangi wrote:
Is your asterisk server behind the firewall or NAT. Does your FS respond to the invite from asterisk, Did you run sip trace on both asterisk and FS. I mean it will be useful to know if the below sip trace is from asterisk or FS. If from asterisk, then need to make sure if FS got the request and how it replied to that request.
Hope this will help you in debugging the problem.
Jai www.didforsale.com *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com"
On Mon, Sep 22, 2008 at 4:01 PM, Noah Silverman <noah@allresearch.com> wrote: Tried that and still doesn't work.
I've attached the SIP INVITE so that maybe you'll see something that gives you a clue.
Also, I don't know if it matters, but the FS server is actually in an off site data center. I'm connecting to it remotely from my office. Works fine for a single polycom phone. (That, and the sound quality is AMAZING! )
Here is what I have in Asterisk now... [Freeswitch] host=111.111.111.111 username=3235551212 secret=password fromdomain=111.111.111.111 port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw
Here's the SIP INVITE. (IP's changed to protect the innocent.) 111.111.111.111 is the address of my FS server 222.222.222.222 is the address of my asterisk server 3235551212 is my username/did/account in FS
U 222.222.222.222:1024 -> 111.111.111.111:5060 INVITE sip:13237773456@111.111.111.111 SIP/2.0. Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK03cd6fd2;rport. From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as146a87d7. To: <sip:13237773456@111.111.111.111>. Contact: <sip:3235551212@222.222.222.222>. Call-ID: 4382446d3a46269d7f469e93147b46eb@111.111.111.111 CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Remote-Party-ID: "3235551212" <sip: 3235551212@111.111.111.111>;privacy=off;screen=no. Proxy-Authorization: Digest username="3235551212", realm="111.111.111.111", algorithm=MD5, uri="sip:13237773456@111.111.111.111 ", nonce="c52ba984-f888-dd11-80f4-00188b37805b", response="01635afacb7eeebc6fc0888991c0411d", qop=auth, cnonce="4e065f27", nc=00000001. Date: Mon, 22 Sep 2008 22:54:14 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 234.
On Sep 22, 2008, at 2:35 PM, Brian West wrote:
you'll need to set from-domain in the sip.conf on asterisk ;)
/b
On Sep 22, 2008, at 2:30 PM, Noah Silverman wrote:
Below is the config in my sip.conf for asterisk. (IP and DID changed for privacy)
[Freeswitch] host=111.111.111.111 username=3235551212 secret=password port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw
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What was the response and where did it go? I need to see the full sip trace along with console output. TPORT_LOG=1 ./freeswitch /b On Sep 22, 2008, at 4:01 PM, Noah Silverman wrote:
U 222.222.222.222:1024 -> 111.111.111.111:5060 INVITE sip:13237773456@111.111.111.111 SIP/2.0. Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK03cd6fd2;rport. From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as146a87d7. To: <sip:13237773456@111.111.111.111>. Contact: <sip:3235551212@222.222.222.222>. Call-ID: 4382446d3a46269d7f469e93147b46eb@111.111.111.111 CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Remote-Party-ID: "3235551212" <sip: 3235551212@111.111.111.111>;privacy=off;screen=no. Proxy-Authorization: Digest username="3235551212", realm="111.111.111.111", algorithm=MD5, uri="sip:13237773456@111.111.111.111 ", nonce="c52ba984-f888-dd11-80f4-00188b37805b", response="01635afacb7eeebc6fc0888991c0411d", qop=auth, cnonce="4e065f27", nc=00000001. Date: Mon, 22 Sep 2008 22:54:14 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 234.
NOTE: IP and DID changed. Freeswitch server = 111.111.111.111 Asterisk server = 222.222.222.222 DID = 3235551212 --------------- recv 862 bytes from udp/[222.222.222.222]:1024 at 03:51:22.604274: ------------------------------------------------------------------------ INVITE sip:13237773456@111.111.111.111 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK38495771;rport From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as68ae65f8 To: <sip:13237773456@111.111.111.111> Contact: <sip:3235551212@10.0.1.100> Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "3235551212" <sip: 3235551212@111.111.111.111>;privacy=off;screen=no Date: Tue, 23 Sep 2008 03:57:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 234 v=0 o=root 3463 3463 IN IP4 10.0.1.100 s=session c=IN IP4 10.0.1.100 t=0 0 m=audio 13808 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 356 bytes to udp/[222.222.222.222]:1024 at 03:51:22.606561: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.100 :5060;branch=z9hG4bK38495771;rport=1024;received=222.222.222.222 From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as68ae65f8 To: <sip:13237773456@111.111.111.111> Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported Content-Length: 0 ------------------------------------------------------------------------ send 833 bytes to udp/[222.222.222.222]:1024 at 03:51:22.636580: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.1.100 :5060;branch=z9hG4bK38495771;rport=1024;received=222.222.222.222 From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as68ae65f8 To: <sip:13237773456@111.111.111.111>;tag=yHjpHt1BmS7yS Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include- session-description, presence.winfo, message-summary Proxy-Authenticate: Digest realm="111.111.111.111", nonce="bacd48e3-2289-dd11-80f4-00188b37805b", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 481 bytes from udp/[222.222.222.222]:1024 at 03:51:22.669730: ------------------------------------------------------------------------ ACK sip:13237773456@111.111.111.111 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK38495771;rport From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as68ae65f8 To: <sip:13237773456@111.111.111.111>;tag=yHjpHt1BmS7yS Contact: <sip:3235551212@10.0.1.100> Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "3235551212" <sip: 3235551212@111.111.111.111>;privacy=off;screen=no Content-Length: 0 ------------------------------------------------------------------------ recv 1119 bytes from udp/[222.222.222.222]:1024 at 03:51:22.670427: ------------------------------------------------------------------------ INVITE sip:13237773456@111.111.111.111 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK59b179c5;rport From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as68ae65f8 To: <sip:13237773456@111.111.111.111> Contact: <sip:3235551212@10.0.1.100> Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "3235551212" <sip: 3235551212@111.111.111.111>;privacy=off;screen=no Proxy-Authorization: Digest username="3235551212", realm="111.111.111.111", algorithm=MD5, uri="sip:13237773456@111.111.111.111 ", nonce="bacd48e3-2289-dd11-80f4-00188b37805b", response="ff3cdb8073becadfe55acfdaf5216441", qop=auth, cnonce="0fe2ecd1", nc=00000001 Date: Tue, 23 Sep 2008 03:57:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 234 v=0 o=root 3463 3464 IN IP4 10.0.1.100 s=session c=IN IP4 10.0.1.100 t=0 0 m=audio 13808 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 356 bytes to udp/[222.222.222.222]:1024 at 03:51:22.673443: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.1.100 :5060;branch=z9hG4bK59b179c5;rport=1024;received=222.222.222.222 From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as68ae65f8 To: <sip:13237773456@111.111.111.111> Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 CSeq: 103 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported Content-Length: 0 ------------------------------------------------------------------------ send 691 bytes to udp/[222.222.222.222]:1024 at 03:51:22.784872: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.0.1.100 :5060;branch=z9hG4bK59b179c5;rport=1024;received=222.222.222.222 From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as68ae65f8 To: <sip:13237773456@111.111.111.111>;tag=ZtBFKNjFH2XHN Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 CSeq: 103 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include- session-description, presence.winfo, message-summary Content-Length: 0 ------------------------------------------------------------------------ recv 481 bytes from udp/[222.222.222.222]:1024 at 03:51:22.811109: ------------------------------------------------------------------------ ACK sip:13237773456@111.111.111.111 SIP/2.0 Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK59b179c5;rport From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as68ae65f8 To: <sip:13237773456@111.111.111.111>;tag=ZtBFKNjFH2XHN Contact: <sip:3235551212@10.0.1.100> Call-ID: 1e4dd5ee5a2fd5d347b551d3714f71e3@111.111.111.111 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "3235551212" <sip: 3235551212@111.111.111.111>;privacy=off;screen=no Content-Length: 0 ------------------------------------------------------------------------ On Sep 22, 2008, at 7:00 PM, Brian West wrote:
What was the response and where did it go? I need to see the full sip trace along with console output.
TPORT_LOG=1 ./freeswitch
/b
On Sep 22, 2008, at 4:01 PM, Noah Silverman wrote:
U 222.222.222.222:1024 -> 111.111.111.111:5060 INVITE sip:13237773456@111.111.111.111 SIP/2.0. Via: SIP/2.0/UDP 10.0.1.100:5060;branch=z9hG4bK03cd6fd2;rport. From: "3235551212" <sip:3235551212@111.111.111.111>;tag=as146a87d7. To: <sip:13237773456@111.111.111.111>. Contact: <sip:3235551212@222.222.222.222>. Call-ID: 4382446d3a46269d7f469e93147b46eb@111.111.111.111 CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Remote-Party-ID: "3235551212" <sip: 3235551212@111.111.111.111>;privacy=off;screen=no. Proxy-Authorization: Digest username="3235551212", realm="111.111.111.111", algorithm=MD5, uri="sip:13237773456@111.111.111.111 ", nonce="c52ba984-f888-dd11-80f4-00188b37805b", response="01635afacb7eeebc6fc0888991c0411d", qop=auth, cnonce="4e065f27", nc=00000001. Date: Mon, 22 Sep 2008 22:54:14 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 234.
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Hello, I'm having trouble getting the accountcode variable to show up in the CDR. Also, the effective_caller_id_number isn't getting passed through to the actual call. (Separate but possible related problem) Calls are working, the CDR looks great, I just don't get these two variables.. Here is an example entry in my directory: <user id="3235551212"> <params> <param name="password" value="1234" /> </params> <variables> <variable name="accountcode" value="432]"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_number" value="3235551212]"/> </variables> </user> And then in my dialplan... <extension name="Vitelity_Outbound"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="bridge" data="sofia/external/$1@outbound1.vitelity.net "/> </condition> </extension> My guess is that maybe I need to export the variable somewhere, or pass it to the actual extension doing the bridging. Anybody have any ideas?? -Noah
those vars will only be set on calls that were authenticated with the user in question you can force the variables to be applied to any channel with the put this right before the bridge line (assuming ${domain} is the domain that user lives in: <action application="set_user" data="3235551212@${domain}"/> On Thu, Sep 25, 2008 at 1:40 PM, Noah Silverman <noah@allresearch.com>wrote:
Hello,
I'm having trouble getting the accountcode variable to show up in the CDR.
Also, the effective_caller_id_number isn't getting passed through to the actual call. (Separate but possible related problem)
Calls are working, the CDR looks great, I just don't get these two variables..
Here is an example entry in my directory:
<user id="3235551212"> <params> <param name="password" value="1234" /> </params> <variables> <variable name="accountcode" value="432]"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_number" value="3235551212]"/> </variables> </user>
And then in my dialplan...
<extension name="Vitelity_Outbound"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="bridge" data="sofia/external/$1@outbound1.vitelity.net "/> </condition> </extension>
My guess is that maybe I need to export the variable somewhere, or pass it to the actual extension doing the bridging.
Anybody have any ideas??
-Noah
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Very interesting! I noticed that there are three similarly-named apps on the wiki that could use some love: set_global set_profile_var set_user Could someone in the know throw some light on these three? I'll be happy to wikify anything that gets posted to the list. -MC On Thu, Sep 25, 2008 at 12:23 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
those vars will only be set on calls that were authenticated with the user in question you can force the variables to be applied to any channel with the
put this right before the bridge line (assuming ${domain} is the domain that user lives in:
<action application="set_user" data="3235551212@${domain}"/>
On Thu, Sep 25, 2008 at 1:40 PM, Noah Silverman <noah@allresearch.com>wrote:
Hello,
I'm having trouble getting the accountcode variable to show up in the CDR.
Also, the effective_caller_id_number isn't getting passed through to the actual call. (Separate but possible related problem)
Calls are working, the CDR looks great, I just don't get these two variables..
Here is an example entry in my directory:
<user id="3235551212"> <params> <param name="password" value="1234" /> </params> <variables> <variable name="accountcode" value="432]"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_number" value="3235551212]"/> </variables> </user>
And then in my dialplan...
<extension name="Vitelity_Outbound"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="bridge" data="sofia/external/$1@outbound1.vitelity.net "/> </condition> </extension>
My guess is that maybe I need to export the variable somewhere, or pass it to the actual extension doing the bridging.
Anybody have any ideas??
-Noah
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
FYI, Here is what I get from sofia status Call-ID 5e7878be778e8d5f04536e0515bd883e@127.0.0.1 User 3235551212@111.111.111.111 Contact "user" <sip:3235551212@222.222.222.222:1024;fs_nat=yes> Agent Asterisk PBX Status Registered(UDP-NAT)(unknown) EXP(2008-09-25 13:12:33) On Sep 25, 2008, at 12:30 PM, Michael Collins wrote:
Very interesting! I noticed that there are three similarly-named apps on the wiki that could use some love: set_global set_profile_var set_user
Could someone in the know throw some light on these three? I'll be happy to wikify anything that gets posted to the list. -MC
On Thu, Sep 25, 2008 at 12:23 PM, Anthony Minessale <anthony.minessale@gmail.com
wrote: those vars will only be set on calls that were authenticated with the user in question you can force the variables to be applied to any channel with the
put this right before the bridge line (assuming ${domain} is the domain that user lives in:
<action application="set_user" data="3235551212@${domain}"/>
On Thu, Sep 25, 2008 at 1:40 PM, Noah Silverman <noah@allresearch.com> wrote: Hello,
I'm having trouble getting the accountcode variable to show up in the CDR.
Also, the effective_caller_id_number isn't getting passed through to the actual call. (Separate but possible related problem)
Calls are working, the CDR looks great, I just don't get these two variables..
Here is an example entry in my directory:
<user id="3235551212"> <params> <param name="password" value="1234" /> </params> <variables> <variable name="accountcode" value="432]"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_number" value="3235551212]"/> </variables> </user>
And then in my dialplan...
<extension name="Vitelity_Outbound"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="bridge" data="sofia/external/$1@outbound1.vitelity.net "/> </condition> </extension>
My guess is that maybe I need to export the variable somewhere, or pass it to the actual extension doing the bridging.
Anybody have any ideas??
-Noah
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
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OK, That makes sense, BUT if the call is coming from a registered user why do I need to explicitly set it again. (If I dial a call from my Polycom phone registered with FS, shouldn't it know which user I am???) Thanks, -N On Sep 25, 2008, at 12:23 PM, Anthony Minessale wrote:
those vars will only be set on calls that were authenticated with the user in question you can force the variables to be applied to any channel with the
put this right before the bridge line (assuming ${domain} is the domain that user lives in:
<action application="set_user" data="3235551212@${domain}"/>
On Thu, Sep 25, 2008 at 1:40 PM, Noah Silverman <noah@allresearch.com> wrote: Hello,
I'm having trouble getting the accountcode variable to show up in the CDR.
Also, the effective_caller_id_number isn't getting passed through to the actual call. (Separate but possible related problem)
Calls are working, the CDR looks great, I just don't get these two variables..
Here is an example entry in my directory:
<user id="3235551212"> <params> <param name="password" value="1234" /> </params> <variables> <variable name="accountcode" value="432]"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_number" value="3235551212]"/> </variables> </user>
And then in my dialplan...
<extension name="Vitelity_Outbound"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="bridge" data="sofia/external/$1@outbound1.vitelity.net "/> </condition> </extension>
My guess is that maybe I need to export the variable somewhere, or pass it to the actual extension doing the bridging.
Anybody have any ideas??
-Noah
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yes that is if you're really authenticating ;) /b On Sep 25, 2008, at 12:46 PM, Noah Silverman wrote:
OK,
That makes sense, BUT if the call is coming from a registered user why do I need to explicitly set it again. (If I dial a call from my Polycom phone registered with FS, shouldn't it know which user I am???)
Thanks,
-N
I thought I was. If I do a "sofia status profile internal" I see myself registered there... How else can I be sure? -N On Sep 25, 2008, at 12:51 PM, Brian West wrote:
Yes that is if you're really authenticating ;)
/b
On Sep 25, 2008, at 12:46 PM, Noah Silverman wrote:
OK,
That makes sense, BUT if the call is coming from a registered user why do I need to explicitly set it again. (If I dial a call from my Polycom phone registered with FS, shouldn't it know which user I am???)
Thanks,
-N
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If you have blind reg on and auth-calls set to false then you'll still show up .. we have to challenge you for those setting to go into effect. So double check the profile for auth-calls and accept-blind-reg. /b On Sep 25, 2008, at 1:07 PM, Noah Silverman wrote:
I thought I was.
If I do a "sofia status profile internal" I see myself registered there...
How else can I be sure?
-N
OK, bind_reg not in my config file at all. auth_calls set to "true" for the default context. -N On Sep 25, 2008, at 1:10 PM, Brian West wrote:
If you have blind reg on and auth-calls set to false then you'll still show up .. we have to challenge you for those setting to go into effect.
So double check the profile for auth-calls and accept-blind-reg.
/b
On Sep 25, 2008, at 1:07 PM, Noah Silverman wrote:
I thought I was.
If I do a "sofia status profile internal" I see myself registered there...
How else can I be sure?
-N
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participants (5)
-
Anthony Minessale
-
Brian West
-
Jai Rangi
-
Michael Collins
-
Noah Silverman