inbound: one-way audio; outbound:two-way audio

benign

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#1
What would cause inbound calls to not be able to hear the caller, but outbound calls have two-way comm?

Setup is like this:

Phone - > Elastix -> Pfsense -> internet

I have the same setup at another location, and everything works fine both ways. Same provider, same phones. Same phone config files, etc.. etc..

I have every single port opened and going to elastix from the external interface (for testing). This doesn't look like a NAT issue as there are no remote phones. When we put remote extensions to call into our first site, I had to configure everything for NAT to get it working. Before that, there was no nat=yes or qualify=no anywhere.

It seems to not be allowing voice packets passed the firewall, or to elastix, or to the phone if any only if the call is inbound. Outbound calls function 100% correctly.
 

dicko

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#2
That is ALWAYS a nat/router/firewall misconfiguration in the router or the Elastix box, it can't be anything else, any inbound udp connections on ports 10000-20000 must be forwarded to the Asterisk box, the asterisk box must nat translate to the external IP if necessary. Check the documentation again, PFSense works perfectly apart from tftp traversal (yet)
 

benign

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#3
Here are details of my setup. Do you see anything out of place?

Telephone -> Elastix -> Firewall -> Internet

Firewall: NAT: Port forward (I've also tried this with *)

If ------- Proto -------- Ext. Port ------ NAT IP ------- Int. Port
WAN TCP/UDP 5004-5082 <elastix box> 5004-5082
WAN TCP/UDP 10000-65535 <elastix box> 10000-65535
WAN UDP 4569 <elastix box> 4569

Firewall: Rules (I've tried this with specific source ports)

Proto --- Source -- Port -- Destination ----- Port ------ GW -- Sch.
TCP/UDP * * <elastix box> 5004-5082 * *
TCP/UDP * * <elastix box> 10000-65535 * *
UDP * * <elastix box> 4569 * *

Elastix Trunk Details:
(These don't require any username/auth). They are set up the same way on another working box.
---------------------
PEER:
host=38.xxx.225.xxx
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833

USER:
context=ext-did
host=38.xxx.225.xxx
insecure=very
type=friend

---------------
Extension: 1000

secret: <secret>
dtmfmode: auto
canreinvite: no
context: from-internal
host: dynamic
type: friend
nat: no
port: 5060
qualify: yes
deny: 0.0.0.0/0.0.0.0
permit 0.0.0.0/0.0.0.0

sip_nat.conf (not using NAT)

externip=72.xx.202.xxx
localnet=10.20.0.0/255.255.255.0

elastix*CLI> sip show peer 1000
elastix*CLI>

* Name : 1000
Secret : <Set>
MD5Secret : <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 1000@default
VM Extension : *97
LastMsgsSent : 0/0
Call limit : 50
Dynamic : Yes
Callerid : "device" <1000>
MaxCallBR : 384 kbps
Expire : 2035
Insecure : no
Nat : RFC3581
ACL : Yes
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : auto
LastMsg : 0
ToHost :
Addr->IP : <phone IP> Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 1000
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing: No
Status : OK (88 ms)
Useragent : Cisco-CP7960G/8.0
Reg. Contact : sip:1000@<phone IP>:5060;transport=udp


elastix*CLI> sip show settings
elastix*CLI>

Global Settings:
----------------
SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
Always auth rejects: Yes
Call limit peers only: Yes
Direct RTP setup: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
T38 fax pt UDPTL: No
RFC2833 Compensation: No
SIP realtime: Disabled

Global Signalling Settings:
---------------------------
Codecs: 0xc (ulaw|alaw)
Codec Order: ulaw:20,alaw:20
T1 minimum: 100
Relax DTMF: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
elastix*CLI>
Default Settings:
-----------------
Context: from-sip-external
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
 

dicko

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#4
I think if your localnet is 10.20.0.0/24 and your externip is static you probably are NATTED and SHOULD be using it in sip*.conf

check with

sip set debug ip 72.xx.202.xxx

to make sure the headers are rewritten properly.
 

benign

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#5
I did a sip debug on a peer and called it. This is what their elastix box CLI gives from the time I call, until I hang up. I can hear them, they can not hear me.

I don't see anything for the external address here, just the address of the phone (.53), and the address of the elastix box (.29)




I did a sip debug on a peer and called it. This is what their elastix box CLI gives from the time I call, until I hang up. I can hear them, they can not hear me.

I don't see anything for the external address here, just the address of the phone (.53), and the address of the elastix box (.29)




-- Executing [81xxxx6086@ext-did:1] Set("SIP/from-321-170-0926c3c8", "__FROM_DID=81xxxx6086") in new stack
-- Executing [81xxxx6086@ext-did:2] Gosub("SIP/from-321-170-0926c3c8", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/from-321-170-0926c3c8", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/from-321-170-0926c3c8", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/from-321-170-0926c3c8", "") in new stack
-- Executing [81xxxx6086@ext-did:3] ExecIf("SIP/from-321-170-0926c3c8", "1 |Set|CALLERID(name)=+1xxx373xx52") in new stack
-- Executing [81xxxx6086@ext-did:4] Set("SIP/from-321-170-0926c3c8", "FAX_RX=") in new stack
-- Executing [81xxxx6086@ext-did:5] Set("SIP/from-321-170-0926c3c8", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [81xxxx6086@ext-did:6] SetCallerPres("SIP/from-321-170-0926c3c8", "allowed_not_screened") in new stack
-- Executing [81xxxx6086@ext-did:7] Goto("SIP/from-321-170-0926c3c8", "from-did-direct|1001|1") in new stack
-- Goto (from-did-direct,1001,1)
-- Executing [1001@from-did-direct:1] Macro("SIP/from-321-170-0926c3c8", "exten-vm|1001|1001") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/from-321-170-0926c3c8", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/from-321-170-0926c3c8", "AMPUSER=+1xxx373xx52") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/from-321-170-0926c3c8", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/from-321-170-0926c3c8", "1|Set|REALCALLERIDNUM=+1xxx373xx52") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/from-321-170-0926c3c8", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/from-321-170-0926c3c8", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/from-321-170-0926c3c8", "1?report") in new stack
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/from-321-170-0926c3c8", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/from-321-170-0926c3c8", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/from-321-170-0926c3c8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/from-321-170-0926c3c8", "Using CallerID "+1xxx373xx52" <+1xxx373xx52>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/from-321-170-0926c3c8", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/from-321-170-0926c3c8", "VMBOX=1001") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/from-321-170-0926c3c8", "EXTTOCALL=1001") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/from-321-170-0926c3c8", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/from-321-170-0926c3c8", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/from-321-170-0926c3c8", "RT=15") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/from-321-170-0926c3c8", "record-enable|1001|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/from-321-170-0926c3c8", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/from-321-170-0926c3c8", "recordingcheck|20100506-112119|1273159279.923") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100506-112119|1273159279.923: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/from-321-170-0926c3c8", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/from-321-170-0926c3c8", "dial|15|tTrwW|1001") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/from-321-170-0926c3c8", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/from-321-170-0926c3c8", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is '+1xxx373xx52' number is '+1xxx373xx52'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 1001 to extension map
> dialparties.agi: Extension 1001 has call screening off
-- dialparties.agi: Extension 1001 cf is disabled
-- dialparties.agi: Extension 1001 do not disturb is disabled
> dialparties.agi: extnum 1001 has: cw: 0; hascfb: 0 [] hascfu: 0 []
dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 1001 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 1001
-- dialparties.agi: dbset CALLTRACE/1001 to +1xxx373xx52
-- dialparties.agi: Filtered ARG3: 1001
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/from-321-170-0926c3c8", "SIP/1001|15|tTrwW") in new stack
Audio is at 10.xx.x.29 port 13516
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.xx.x.53:5060:
INVITE sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK199ee84f;rport
From: "+1xxx373xx52" <sip:+1xxx373xx52@10.xx.x.29>;tag=as2c40a970
To: <sip:1001@10.xx.x.53:5060;transport=udp>
Contact: <sip:+1xxx373xx52@10.xx.x.29>
Call-ID: 42bf4e872bb883d9628cff865315cb8a@10.xx.x.29
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 06 May 2010 15:21:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 3066 3066 IN IP4 10.xx.x.29
s=session
c=IN IP4 10.xx.x.29
t=0 0
m=audio 13516 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv

---
-- Called 1001
-- SIP/1001-092f3cd0 is ringing
Found RTP audio format 0
Peer audio RTP is at port 10.xx.x.53:25046
Found audio description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.xx.x.53:25046
list_route: hop: <sip:1001@10.xx.x.53:5060;transport=udp>
set_destination: Parsing <sip:1001@10.xx.x.53:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.xx.x.53, port 5060
Transmitting (no NAT) to 10.xx.x.53:5060:
ACK sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK5e476eba;rport
From: "+1xxx373xx52" <sip:+1xxx373xx52@10.xx.x.29>;tag=as2c40a970
To: <sip:1001@10.xx.x.53:5060;transport=udp>;tag=000f8f7608ac048a21c17a9d-33e5dd6f
Contact: <sip:+1xxx373xx52@10.xx.x.29>
Call-ID: 42bf4e872bb883d9628cff865315cb8a@10.xx.x.29
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/1001-092f3cd0 answered SIP/from-321-170-0926c3c8
-- Executing [h@macro-dial:1] Macro("SIP/from-321-170-0926c3c8", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/from-321-170-0926c3c8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/from-321-170-0926c3c8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/from-321-170-0926c3c8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/from-321-170-0926c3c8", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/from-321-170-0926c3c8' in macro 'hangupcall'
== Spawn h extension (macro-dial, h, 1) exited non-zero on 'SIP/from-321-170-0926c3c8'
Scheduling destruction of SIP dialog '42bf4e872bb883d9628cff865315cb8a@10.xx.x.29' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:1001@10.xx.x.53:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.xx.x.53, port 5060
Reliably Transmitting (no NAT) to 10.xx.x.53:5060:
BYE sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK1f8edd1a;rport
From: "+1xxx373xx52" <sip:+1xxx373xx52@10.xx.x.29>;tag=as2c40a970
To: <sip:1001@10.xx.x.53:5060;transport=udp>;tag=000f8f7608ac048a21c17a9d-33e5dd6f
Call-ID: 42bf4e872bb883d9628cff865315cb8a@10.xx.x.29
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/from-321-170-0926c3c8' in macro 'dial'
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/from-321-170-0926c3c8' in macro 'exten-vm'
== Spawn extension (from-did-direct, 1001, 1) exited non-zero on 'SIP/from-321-170-0926c3c8'
Really destroying SIP dialog '42bf4e872bb883d9628cff865315cb8a@10.xx.x.29' Method: INVITE
 

dicko

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#6
Did you try either in sip*.conf

nat=1

or
nat=route

?
 

benign

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#7
I've tried nat=yes, it show in the sip debug that it is transmitting with nat after a sip reload. I just tried nat = route. Sip details show nat: always instead of rfc3581. With nat=route or nat=yes, there is no audio both ways :(
 

dicko

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#8
try the nat=route

also

rtp debug ip <voip provider>

To watch the rtp traffic in asterisk (It will presumably be one way) and at the same time

tcpdump -nn host <voip provider>

from bash to see where the missing rtp packts are being sent under the various nat=? settings
 

dicko

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#9
I have no problems with pfsense, but use 1:1 you might want to set your NAT outbound route to manual and use static port mapping
 

benign

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#10
[root@elastix ~]# tcpdump -nn host 38.xxx.xxx.170
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes
13:55:27.218598 IP 38.xxx.xxx.170.5060 > 10.xx.x.29.5060: SIP, length: 1109
13:55:27.219325 IP 10.xx.x.29.5060 > 38.xxx.xxx.170.5060: SIP, length: 699
13:55:27.432059 IP 10.xx.x.29.5060 > 38.xxx.xxx.170.5060: SIP, length: 715
13:55:30.796670 IP 10.xx.x.29.5060 > 38.xxx.xxx.170.5060: SIP, length: 1005
13:55:30.977776 IP 38.xxx.xxx.170.5060 > 10.xx.x.29.5060: SIP, length: 680
13:55:38.587758 IP 38.xxx.xxx.170.5060 > 10.xx.x.29.5060: SIP, length: 672
13:55:38.587988 IP 10.xx.x.29.5060 > 38.xxx.xxx.170.5060: SIP, length: 564

This is weird because the trunk is actually set up to be 38.xxx.xxx.190. It's not showing
anything going out to .190 so I assume whatever is sent to 190 just comes back through 170.

I testing changing the incoming peer settings to 170, which had no effect.

RTP Debug and SIP headers information for a phone call to the server is this:







INVITE sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK3248a0d8;rport
From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xx.x.29>;tag=as35702444
To: <sip:1001@10.xx.x.53:5060;transport=udp>
Contact: <sip:+1xxxxxxxxxx@10.xx.x.29>
Call-ID: 4368d3db6f38a5d20ffbcec65f2abf16@10.xx.x.29
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 06 May 2010 18:04:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 3066 3066 IN IP4 10.xx.x.29
s=session
c=IN IP4 10.xx.x.29
t=0 0
m=audio 13722 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv

---
-- Called 1001
-- SIP/1001-092b3200 is ringing
Found RTP audio format 0
Peer audio RTP is at port 10.xx.x.53:17922
Found audio description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.xx.x.53:17922
list_route: hop: <sip:1001@10.xx.x.53:5060;transport=udp>
set_destination: Parsing <sip:1001@10.xx.x.53:5060;transport=udp> for address/port to send to
set_destination: set destination to 10.xx.x.53, port 5060
Transmitting (no NAT) to 10.xx.x.53:5060:
ACK sip:1001@10.xx.x.53:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.xx.x.29:5060;branch=z9hG4bK4f3d44ad;rport
From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xx.x.29>;tag=as35702444
To: <sip:1001@10.xx.x.53:5060;transport=udp>;tag=000f8f7608ac05476eb02cc5-1c7876ab
Contact: <sip:+1xxxxxxxxxx@10.xx.x.29>
Call-ID: 4368d3db6f38a5d20ffbcec65f2abf16@10.xx.x.29
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
-- SIP/1001-092b3200 answered SIP/from-321-b6a4ee38
Got RTP packet from 10.xx.x.53:17922 (type 00, seq 003330, ts 533312, len 000160)
Sent RTP packet to 199.173.105.141:31934 (type 00, seq 021818, ts 533312, len 000160)
Got RTP packet from 10.xx.x.53:17922 (type 00, seq 003331, ts 533472, len 000160)
Sent RTP packet to 199.173.105.141:31934 (type 00, seq 021819, ts 533472, len 000160)

199.173.105.141 is some verizon equipment.. which is the ISP the site I'm working with is using.
 

dicko

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#11
I would try something else but pfsense at this time, if it works go to the pfsense forum with this problem.

sorry I can't be of more help

dicko
 

benign

Joined
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#12
Ok, well I appreciate the help.
 

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