Displaying 20 results from an estimated 29 matches for "vrtp".
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rtp
2010 Jan 29
1
callerid not working over sip
...nternal:3] Set("DAHDI/1-1", "CALLERID="Test"
<447>") in new stack
-- Executing [170 at internal:4] Dial("DAHDI/1-1",
"SIP/office-home-sip/170") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called office-home-sip/170
On the office asterisk:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL TOS bits 184
==...
2009 Jul 03
0
e164.org and tollfree ENUM records
...xecuting [18002662278 at outbound:7] Exec("SIP/aastra-sip1-0c004d98",
"Dial(SIP/164164180018002662278 at sip.tollfreegateway.com,40,KL(7200000:120000)T)")
in new stack
-- Limit Data for this call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 164164180018002662278 at sip.tollfreegateway.com
-- Got SIP response 480 "Temporarily Unavailable" back from 204.8.45.222
-- SIP/sip.tollfreegateway.com-140f2228 is...
2009 Nov 22
1
transferring SIP call: no voice
...rt forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls from Junction and Teliax.
CLI on A looks right:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Executing [3008384e0 at sipgate-test:1]
Answer("SIP/sipgate-00000016", "") in new stack
-- Executing [3008384e0 at sipgate-test:2]
Goto("SIP/sipgate-00000016", "home,447,1")...
2010 Dec 20
5
DIALSTATUS on CANCEL
...=> s-BUSY,1, NoOp()
exten => s-BUSY,n, Return()
This is what we get on a BUSY call:
-----------------------------------
-- Executing [11111111 at incoming-private:3] Dial("SIP/Proxy-0000002b",
"SIP/1001,50") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called 1001
-- Got SIP response 486 "Busy Here" back from 10.0.0.1
-- SIP/1001-0000002c is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [11111111 at incoming-private:4] NoOp("SIP/Proxy-0000002b"...
2009 Jun 26
0
Problem with RetryDial
...cfmc_cdi_private:2]
RetryDial("Local/retry_number at cfmc_cdi_private-4ff4;2",
"another-time,10,7,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQueue^SIP/
GXP280_18)") in new stack
[2009-06-26 08:12:47.898] == Using SIP RTP CoS mark 5
[2009-06-26 08:12:47.898] == Using SIP VRTP CoS mark 6
[2009-06-26 08:12:47.899] -- Called GXP280_18
[2009-06-26 08:12:47.942] -- SIP/GXP280_18-09e5ea20 is ringing
[2009-06-26 08:12:57.900] -- Nobody picked up in 10000 ms
[2009-06-26 08:12:57.901] --
<Local/retry_number at cfmc_cdi_private-4ff4;2> Playing 'another-t...
2005 May 20
0
Registering with second SIP service causes error every 2 seconds - what is going on?
...:33 VERBOSE[5520]: Asterisk Ready.
May 20 10:01:33 DEBUG[5520]: (Provisional) Stopping retransmission (but
retaining packet) on '2657c93c79b12ba855365135794f06e5@172.31.254.2'
Request 102: Found
May 20 10:01:33 DEBUG[5520]: Setting NAT on RTP to 0
May 20 10:01:33 DEBUG[5520]: Setting NAT on VRTP to 0
May 20 10:01:33 DEBUG[5520]: ##### Testing 172.31.250.5 with 172.31.0.0
May 20 10:01:33 DEBUG[5520]: Stopping retransmission on
'2657c93c79b12ba855365135794f06e5@172.31.254.2' of Request 102: Found
May 20 10:01:33 DEBUG[5520]: Setting NAT on RTP to 0
May 20 10:01:33 DEBUG[5520]: Settin...
2010 Mar 26
2
dnd not working correctly
...ave posted this question couple of times and never really got any hits i wasn't able to provide any debug info
Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)
Verbosity is at least 4
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Extension Changed 117[ext-local] new state InUse for Notify User 102
== Extension Changed 117[ext-local] new state InUse for Notify User 103
== Extension Changed 117[ext-local] new state InUse for Notify User 114
-- Executing [*76 at fro...
2010 Mar 26
1
problem with polarity reverse
...93 00:00:29 (None)
Show not bridged but conversation was normal poth sides everything hear
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2894 do_setnat: Setting NAT on RTP to Off
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2899 do_setnat: Setting NAT on VRTP to Off
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2904 do_setnat: Setting NAT on UDPTL to Off
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:4797 sip_alloc: Allocating new SIP dialog for 842ada2a-77b3dff at 192.168.xx.xx - INVITE (With RTP)
[Mar 26 14:36:38] DEBUG[12001]: chan_sip.c:2894 do_setnat: Set...
2010 Apr 08
1
Linksys/Sipura SPA-3201 FXO/FSA with Asterisk
...en I restart Asterisk,
everything works perfectly. I let Asterisk sit for an hour or so, and it
stops allowing calls to be routed into the assigned extension. I do see
stuff from the communications, at the time the call lands on the
Asterisk server:
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
The logic is that the SPA is registered as an extension on my system,
and incoming calls are routed into the system VIA that extension. The
dialplan that the SPA connects to is:
[gw8028]
exten => 8028,1,Answer
exten => 8028,n,Set(CallerNum=${CALLERID(num)})...
2009 Jul 21
1
Dialplan step that I do not have
...Verbose(2,Doing custom record - before hangup)
[pbx_config]
16. hangup()
[pbx_config]
When call this extension I see this:
[2009-07-21 07:48:43.897] == Using SIP RTP TOS bits 184
[2009-07-21 07:48:43.897] == Using SIP RTP CoS mark 5
[2009-07-21 07:48:43.897] == Using SIP VRTP TOS bits 136
[2009-07-21 07:48:43.897] == Using SIP VRTP CoS mark 6
[2009-07-21 07:48:44.004] -- Executing [*9901 at empl:1]
Playback("SIP/dickenson-174c2010", "vm-goodbye") in new stack
[2009-07-21 07:48:44.120] -- <SIP/dickenson-174c2010> Playing
'vm-goodby...
2009 Nov 13
2
openSuse 11.2 and dahdi-linux
OK, I know it's only just out today but this is what I get when
compiling dahdi-linux.
make -C drivers/dahdi/firmware firmware-loaders
make[1]: Entering directory
`/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware'
make[1]: Leaving directory
`/usr/src/dahdi-linux-2.2.0.2/drivers/dahdi/firmware'
make -C /lib/modules/2.6.31.5-0.1-default/build
2010 Mar 03
0
CALLERID(num) not working
...Asterisk 1.6.0.13.
Here are the two calls as shown on the CLI of the asterisk box with the PRI line. The first works and the second does not.
[2010-03-02 13:32:09.520] == Using SIP RTP TOS bits 184
[2010-03-02 13:32:09.520] == Using SIP RTP CoS mark 5
[2010-03-02 13:32:09.520] == Using SIP VRTP TOS bits 136
[2010-03-02 13:32:09.520] == Using SIP VRTP CoS mark 6
[2010-03-02 13:32:09.617] -- Executing [91111111111 at context:1] Set("SIP/username-114ffe50", "MyChan=SIP") in new stack
[2010-03-02 13:32:09.617] -- Executing [91111111111 at context:2] GotoIf("...
2005 Jan 10
0
Problems calling between two local SIP extensions
...-- Executing Dial("IAX2/30@30/2",
"SIP/16|30|tr") in new stack
Jan 8 18:17:55 DEBUG[11919]: SIMPLE DIAL (NO URL)
Jan 8 18:17:55 DEBUG[11919]: Allocating new SIP call for (null)
Jan 8 18:17:55 DEBUG[11919]: Setting NAT on RTP to 0
Jan 8 18:17:55 DEBUG[11919]: Setting NAT on VRTP to 0
Jan 8 18:17:55 DEBUG[11919]: ##### Testing 193.77.90.224 with 192.168.0.0
Jan 8 18:17:55 DEBUG[11919]: Target address 193.77.90.224 is not local,
substituting externip
Jan 8 18:17:55 DEBUG[11919]: Outgoing Call for 16
Jan 8 18:17:55 DEBUG[11919]: Call from user '16' is 1 out of 1
J...
2006 Jun 26
2
1.2.9.1 SIP/Local/Queue behaviours weird
...onds' talk). It is weird.
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Executing
Dial("Local/7188@from-internal-7036,2", "SIP/7188|30|trWwT") in new
stack
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on RTP to 0
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Setting NAT on VRTP to 0
Jun 26 16:53:35 DEBUG[8290] chan_sip.c: Outgoing Call for 7188
Jun 26 16:53:35 VERBOSE[8290] logger.c: -- Called 7188
Jun 26 16:53:35 VERBOSE[8287] logger.c: --
Local/7188@from-internal-7036,1 is ringing
Jun 26 16:53:35 DEBUG[2966] chan_sip.c: (Provisional) Stopping
retransmission (but retaini...
2003 Dec 10
0
Native Bridging and Polycom 600 Solved
...us invite. However, the
Polycom phone tracks its transactions this way - the branch numbers must be
different for new invites. So here's the change:
In chan_sip.c, in transmit_reinvite_with_sdp():
static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp,
struct ast_rtp *vrtp)
{
struct sip_request req;
if (p->canreinvite == REINVITE_UPDATE)
reqprep(&req, p, "UPDATE", 0);
else {
// BEGIN POLYCOM CHANGE
p->branch++;
snprintf(p->via, sizeof(p->via), "SIP/2.0/UDP
%s:%d;branch=z9hG4bK%08x", inet_ntoa(p->ourip),...
2009 Nov 16
1
1.6.0.18-rc3: SendFAX causes restart
...Started /var/log/asterisk/event_log
[Nov 15 19:01:01] VERBOSE[17043] logger.c: Asterisk Dynamic Loader
Starting:
From CLI:
-- Attempting call on SIP/fax at nhi-riverside-sip for s at fax-tx-test:1
(Retry 1)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Executing [s at fax-tx-test:1]
NoOp("SIP/nhi-riverside-sip-00000000", "Context fax-tx-test") in new stack
-- Executing [s at fax-tx-test:2]
System("SIP/nhi-riverside-sip-00000000", &q...
2009 Dec 27
0
Parking function problem ?
...s :
-- Executing [0383824377 at local:1] Wait("SIP/*15-0849ea88", "0")
in new stack
-- Executing [0383824377 at local:2] Dial("SIP/*15-0849ea88", "SIP/
0383xxxxxxx at trunk_sip_2,0,TK") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Called 0383824377 at trunk_sip_2
-- SIP/trunk_sip_2-084a5e18 is ringing
-- SIP/trunk_sip_2-084a5e18 is making progress passing it to SIP/
*15-0849ea88
-- SIP/trunk_sip_2-084a5e18 answered SIP/*15-0849ea88
callbox*CLI>
callbox*CLI>
callbox*CLI>
-- Sta...
2010 May 03
0
Parking problem with outgoing calls
...Here are the logs :
-- Executing [03838xxxxx at local:1] Wait("SIP/*15-0849ea88", "0")
in new stack
-- Executing [03838xxxxx at local:2] Dial("SIP/*15-0849ea88", "SIP/
0383xxxxxxx at trunk_sip_2,0,TK") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Called 0383824377 at trunk_sip_2
-- SIP/trunk_sip_2-084a5e18 is ringing
-- SIP/trunk_sip_2-084a5e18 is making progress passing it to SIP/
*15-0849ea88
-- SIP/trunk_sip_2-084a5e18 answered SIP/*15-0849ea88
callbox*CLI>
callbox*CLI>
callbox*CLI>
-- Started music on hold, cl...
2009 Jul 27
1
disposition "answered" after authenticate??????????
...IP/8000-b51e04e0", "1?1-dial,1") in new stack
-- Goto (macro-trunkdial-failover-0.4,1-dial,1)
-- Executing [1-dial at macro-trunkdial-failover-0.4:1]
Dial("SIP/8000-b51e04e0", "SIP/test/0511111111") in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
== Using UDPTL CoS mark 5
-- Called test/05323939196
-- SIP/Superonline-b474f218 is making progress passing it to
SIP/8000-b51e04e0
-- SIP/Superonline-b474f218 is making progress passing it to
SIP/8000-b51e04e0
== Spawn extension (macro-trunkdial-failover-0.4, 1-dial, 1)...
2010 Jun 11
7
How to stop intruder from registering sip?
...n 10 15:51:20] NOTICE[1662] chan_sip.c: Received SIP subscribe for
peer without mailbox: 151
[Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using SIP RTP TOS bits 184
[Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using SIP RTP CoS mark 5
[Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using SIP VRTP CoS mark 6
[Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using UDPTL TOS bits 184
[Jun 10 15:51:21] VERBOSE[1662] netsock.c: == Using UDPTL CoS mark 5
[Jun 10 15:51:22] VERBOSE[4780] pbx.c: -- Executing
[01125240212154 at longdistance:1] Answer("SIP/151-000000ae", "") i...