[asterisk-users] Problem with SIP-TLS
Hi list! I'm trying to configure my Asterisk to accept SIP-TLS connections, too. I followed this HowTo: http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ But as soon I try to connect to my Asterisk using SIP-TLS I get on Asterisk-CLI: == Problem setting up ssl connection: error:140760FC:lib(20):func(118):reason(252) [Jun 5 20:16:25] WARNING[20826]: tcptls.c:669 handle_tcptls_connection: FILE * open failed! And of course it does NOT connect... Any idea? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP-TLS
ricky gutierrez xserverli...@gmail.com schrieb: compilation problems with the module srtp , check the module module show like srtp Now available on OpenWRT... :( Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP-TLS
2015-06-05 14:29 GMT-06:00 Luca Bertoncello lucab...@lucabert.de: I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucab...@lucabert.de) compilation problems with the module srtp , check the module module show like srtp -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP-TLS
ricky gutierrez xserverli...@gmail.com schrieb: Hi lucas , dou you try this: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Tested right now. Same problem... I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP-TLS
2015-06-05 12:21 GMT-06:00 Luca Bertoncello lucab...@lucabert.de: Hi list! I'm trying to configure my Asterisk to accept SIP-TLS connections, too. I followed this HowTo: http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ But as soon I try to connect to my Asterisk using SIP-TLS I get on Asterisk-CLI: == Problem setting up ssl connection: error:140760FC:lib(20):func(118):reason(252) [Jun 5 20:16:25] WARNING[20826]: tcptls.c:669 handle_tcptls_connection: FILE * open failed! And of course it does NOT connect... Any idea? Thanks Luca Bertoncello (lucab...@lucabert.de) -- Hi lucas , dou you try this: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial regardss -- rickygm http://gnuforever.homelinux.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP 480 from ITSP
I am using voip with Vodafone as SIP peer for outbound telephony and i have a huge problem establishing calls to other people. It works like in 1 of 5 tries. The peer is sending SIP 480 temporarily not available. It took a while to identify this, because on the phone you just hear busy tone. On inbound calls i have not detected problems yet. Calling to mobile numbers works better than to home/office numbers. If i use a different SIP Peer in same dialplan it works without any problem. If i connect a all in one voip router with vodafone peer it works without problem too. Are they lowering availability if they detect Asterisk? Are there any IDs that can be changed, so they cannot detect the PBX ? Are there any settings in Asterisk which affect timeouts or stuff like that? Here: http://voicent.com/kb/index.php/support/autodialer/581/sip-error-480-temporarily-unavailable they write: Depending on the exact cause of the error, your solution may vary. For example, you can try to add another VOIP account for different lines (from the same VOIP service or different one), or slow down the calling (by defining dialing intervals in Voicent gateway) What they mean with slow down and dial interval? I have also the impression that the ISP resonds very quick with the error. Or has it even to do with the router itself? (i am using OpenWRT and had a problem once with NAT-T after public ip change). If i turn on qualify, the peer refuses also to answer (SIP read 403 forbidden). I was wondering, how Asterisk even knows that it is registered? The register and peer status in Asterisk are both ok, i even monitor that in Nagios. Its a big problem in Germany because the ITSP can force you by law to use their own branded router models, which of course dont use Asterisk. So if i would call them, the first answer would be, use a different router. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf
#include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf
type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf
#include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf
type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN})
exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that point
from SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1
s=pjmedia
c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request -
nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=172.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 ---
ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username=6110, realm=asterisk,
nonce=16883b72, uri=sip:7444@172.17.0.17,
response=b75389c5938b4f185b3d31bd4463abf3, algorithm=MD5
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1
s=pjmedia
c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Does each box show up in the others SIP SHOW PEERS?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request - nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 --- ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username=6110, realm=asterisk,
nonce=16883b72, uri=sip:7444@172.17.0.17,
response=b75389c5938b4f185b3d31bd4463abf3, algorithm=MD5
Content-Type: application/sdp
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that
point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4
172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request -
nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog
'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 --- ACK sip:7444@172.17.0.17
SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username=6110, realm
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Looks like a connectivity issue, doesn't it?
IP of box2, 172.17.145.145, doesn't show up even once in the SIP dialogues.
What happens on box2 (asterisk -vvvr and tcpdump port 5060) in the
moment that you place a call through box1 to box2?
Also what's strange is that you are trying to call from box2 to box2?
Because local_SIP is the context on box2, and on box1 it's adhearsion.
The console message you pasted shows @local_SIP however, so it looks
like you are calling from box2 to box2?
Am 10.12.2012 22:53, schrieb Ken D'Ambrosio:
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that
point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request - nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 --- ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Hi, Ken
I have almost the same setup as yours: new
asterisk-SIP-Trixbox(Asterisk 1.4)---PRIpots
Here are my configs:
new box sip.conf:
[126]
directmedia=no
type=friend
host=trixbox_IP_addr
secret=my_secret
username=126 ;this is for outgoing calls from new asterisk via trixbox
fromuser=126 ;this is for outgoing calls from new asterisk via trixbox
context=default
disallow=all
allow=alaw
allow=ulaw
qualify=yes
qualifyfreq=60
nat=yes
pickupgroup=1
callgroup=1
trixbox
[126]
type=friend
secret=mysecret
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=
host=dynamic
dtmfmode=rfc2833
dial=SIP/126
context=from-internal
canreinvite=no
callgroup=
callerid=device 126
accountcode=
call-limit=50
New box's account (126) registers to the Trixbox so as to make incoming calls
from trixbox to new box possible.
The config in the new box implies that the trixbox require authorization in
calls from the new box (username and fromuser options are necessary for this).
Actually looking through the sip.conf in 1.8 asterisk I found that there are
auth option as well as remotesecret and remoteuser - but I can not
understand how they work in case if I need to authorise my outgoing calls
(probably sip.conf will be more logical in the future 12th version).
Hope this helps.
Dmitry Pavlenko
From: Ken D'Ambrosio k...@jots.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 11, 2012 3:53 AM
Subject: Re: [asterisk-users] Problem with SIP trunk I've set up between two *
boxes.
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that
point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4
172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request -
nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0
[asterisk-users] Problem with SIP phone outside local network
I am having a strange problem with an external SIP phone. It can register and receive calls but it cannot initiate any calls. A softphone on the same network works without problems. As far as I can notice the difference is that the hard phone is not sending the proper contact info. In the fullcontact field I can see its private IP address sip:1008@192.168.2.18:5060^3Btransport=udp while the softphone provides the public IP. The hard phone is an Aastra 6730i. A similar phone can make and receive calls when connected from another external network so I do not think it is an Aastra issue. Asterisk is behind a NAT (on DMZ) and has the proper externhost. The sip phone definition has nat=yes Any ideas? Here is a sip debug of the failed call: --- SIP read from UDP:201.141.67.189:41528 --- INVITE sip:1...@pbxwbu.x.org:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6 Max-Forwards: 70 From: 1008 sip:1...@pbxwbu.x.org:5060;tag=fc268bfd9f To: sip:1...@pbxwbu.x.org:5060;user=phone Call-ID: 9567bd1f0b345f53 CSeq: 29187 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: 1008 sip:1008@192.168.2.18:5060;transport=udp; +sip.instance=urn:uuid:--1000-8000-00085D21B027 Supported: path, 100rel, replaces User-Agent: Aastra 6730i/3.2.2.1136 Content-Type: application/sdp Content-Length: 595 v=0 o=MxSIP 0 1 IN IP4 192.168.2.18 s=SIP Call c=IN IP4 192.168.2.18 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv - --- (14 headers 25 lines) --- Sending to 201.141.67.189:41528 (NAT) Using INVITE request as basis request - 9567bd1f0b345f53 Found peer '1008' for '1008' from 201.141.67.189:41528 --- Reliably Transmitting (NAT) to 201.141.67.189:41528 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6;received=201.141.67.189;rport=41528 From: 1008 sip:1...@pbxwbu.x.org:5060;tag=fc268bfd9f To: sip:1...@pbxwbu.x.org:5060;user=phone;tag=as383d0a46 Call-ID: 9567bd1f0b345f53 CSeq: 29187 INVITE Server: Asterisk PBX 1.8.9.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=553cdc38 Content-Length: 0 Scheduling destruction of SIP dialog '9567bd1f0b345f53' in 7040 ms (Method: INVITE) Retransmitting #1 (NAT) to 201.141.67.189:41528: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6;received=201.141.67.189;rport=41528 From: 1008 sip:1...@pbxwbu.x.org:5060;tag=fc268bfd9f To: sip:1...@pbxwbu.x.org:5060;user=phone;tag=as383d0a46 Call-ID: 9567bd1f0b345f53 CSeq: 29187 INVITE Server: Asterisk PBX 1.8.9.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=553cdc38 Content-Length: 0 --- --- SIP read from UDP:201.141.67.189:41528 --- ACK sip:1...@pbxwbu.x.org:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6 Max-Forwards: 70 From: 1008 sip:1...@pbxwbu.x.org:5060;tag=fc268bfd9f To: sip:1...@pbxwbu.x.org:5060;user=phone;tag=as383d0a46 Call-ID: 9567bd1f0b345f53 CSeq: 29187 ACK User-Agent: Aastra 6730i/3.2.2.1136 Content-Length: 0 - --- (9 headers 0 lines) --- --- SIP read from UDP:201.141.67.189:41528 --- ACK sip:1...@pbxwbu.x.org:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bKa522fd15e489606f6 Max-Forwards: 70 From: 1008 sip:1...@pbxwbu.x.org:5060;tag=fc268bfd9f To: sip:1...@pbxwbu.x.org:5060;user=phone;tag=as383d0a46 Call-ID: 9567bd1f0b345f53 CSeq: 29187 ACK User-Agent: Aastra 6730i/3.2.2.1136 Content-Length: 0 - --- (9 headers 0 lines) --- == Using SIP RTP CoS mark 5 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
Hi, thanks a lot by the answers. But without the application Answer() the problem remains. Realized over a battery of tests and refined the problem. Follows: A = External link that came with my Voip number. B = Operator. C = The extent to which A want to speak. A called my number and B answer. If B try to transfer with blindxfer (#) to C works fine. But if B try to transfer with atxfer (*2) he can talk to C, only when B hangs up to complete the transfer begins to generate those warnings on the cli. After the transfer using C atxfer not hear A, but A hears C. I believe it has become clearer now. And as he said, with any codec, and only when the person connects to my VoIP trunks. I did the test with the analogue trunks and atxfer worked normal. Thanks, Rodrigo Lang. 2010/7/20 Stefan Schmidt s...@sil.at Rodrigo Lang schrieb: Good afternoon list. I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the following warning: [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) Detail, this happens with both the codec gsm, ulaw, alaw and g729 and with any of my SIP carrier's (I own three). And only happens when the call is transferred. Does anyone have any idea what could be? Thanks, Rodrigo Lang. hello rodrigo, this is exactly the problem i had. Have a look at issue 17641 (https://issues.asterisk.org/view.php?id=17641) There is a patch for asterisk 1.6.2.9 but its only a single row so you could easy find the position in app_dial.c to patch it by your own. the problem only occurs when you use answer in your dialplan. without an answer this wont happen. best regards. steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
Hi! I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. this is exactly the problem i had. Have a look at issue 17641 (https://issues.asterisk.org/view.php?id=17641) Also look here: https://issues.asterisk.org/view.php?id=17007 Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP
Good afternoon list. I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the following warning: [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) Detail, this happens with both the codec gsm, ulaw, alaw and g729 and with any of my SIP carrier's (I own three). And only happens when the call is transferred. Does anyone have any idea what could be? Thanks, Rodrigo Lang. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
Hi! client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. I *think* there is/was an entry in the bug tracker on this. You might want to search https://issues.asterisk.org (also look for RTP issues with SSRC) and in the meanwhile you could reveal which version of Asterisk you are using. :) Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
This is the exit of core show version: Asterisk 1.6.0.28 built by root @ AST on a i686 running Linux on 2010-06-28 12:21:24 UTC Obg, Rodrigo Lang. 2010/7/20 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Hi! client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. I *think* there is/was an entry in the bug tracker on this. You might want to search https://issues.asterisk.org (also look for RTP issues with SSRC) and in the meanwhile you could reveal which version of Asterisk you are using. :) Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP
Rodrigo Lang schrieb: Good afternoon list. I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the following warning: [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) [Jul 19 14:46:24] WARNING [9220]: chan_sip.c: 5340 sip_write: Asked to transmit frame type 64, while native formats is 0x2 (gsm) (2) read / write = 0x40 (slin) (64) / 0x2 (gsm) (2) Detail, this happens with both the codec gsm, ulaw, alaw and g729 and with any of my SIP carrier's (I own three). And only happens when the call is transferred. Does anyone have any idea what could be? Thanks, Rodrigo Lang. hello rodrigo, this is exactly the problem i had. Have a look at issue 17641 (https://issues.asterisk.org/view.php?id=17641) There is a patch for asterisk 1.6.2.9 but its only a single row so you could easy find the position in app_dial.c to patch it by your own. the problem only occurs when you use answer in your dialplan. without an answer this wont happen. best regards. steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP Subscription Status
Hello All, I've been having some intermittent trouble with an Asterisk 1.2.10 installation that is supporting roughly 50 SIP clients on a LAN, mostly soft phones and about 10 snom VoIP phones. We have a custom soft phone client which displays presence information for various extensions. Unfortunately, this information regularly gets out of sync with the actual status of the various extensions. An extension will show up as 'InUse' or 'Unavailable' when the individual is in fact not on the line, i.e. the status should be 'Idle'. I can verify this by issuing a 'sip show subscriptions' and typically for every client subscribed to the problem extension the status column displays 'InUse'/'Unavailable'. However I have also noticed that in some instances, half the subscribed clients will get an 'Idle' status and the other half will have 'InUse' or 'Unavailable'. Often this behavior will follow some other SNAFU, e.g. a rogue mpg123 process for MoH consuming abnormal amounts of CPU and creating high loads. However, this is not always the case. The lack of a pattern, and more then anything, a simple solution has forced my hand and I'm appealing to the list for help. I've been over voip-info countless times and have searched around more then I care to remember. Restarting the soft phones does nothing to alleviate the problem (which doesn't entirely surprise me as I'm starting to think it's a server side issue). Restarting asterisk itself generally resolves things however this is not an option in the middle of the day. Thanks in advance for all your help. -- Zach Segal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with SIP Subscription Status
Hach Segal a écrit : Hello All, I've been having some intermittent trouble with an Asterisk 1.2.10 Before anything else did you tried an updated asterisk 1.2 The last one is 1.2.28 or something like that, and there has been a lot of security patches, and fixes since your version. Did you look through the changelog / bugs tracker to see if your problem has already been reported ? -- Benoit Plessis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP, attended transfer and GROUP_COUNT
Hi,
maybe someone can give me a hint to solve the following issue. I want to
limit the calls to a specific SIP-destination. Disabling callwaiting at
the phones is not an option, because it should be configured via the *
database.
My solution uses GROUP_COUNT, which works fine most of the time. In case
of attended transfer (on SIP-basis, not via the #-mechanism of asterisk)
I have problems.
To simplfy the scenario I stripped down the dialplan to the following.
From somewhere on the wiki I am using the following context:
exten = 200,1,Set(GROUP()=${CALLERID(num)})
exten = 200,n,GotoIf($[${GROUP_COUNT(${EXTEN})} = 1]?BLOCK)
exten = 200,n,Set(OUTBOUND_GROUP=${EXTEN})
exten = 200,n,Dial(SIP/katrin)
exten = 200,n(BLOCK),Busy
This block is used for other extensions 100 and 150 respectivily. It
works fine until I am using attended transfer.
Example: kwe (Extension 100) is calling katrin (Extension 200). katrin
sets the call on hold and talks to hans (Extension 150).
At the cli I get the following result:
pbxtest*CLI group show channels
ChannelGroup Category
SIP/kwe-081bf188 100 (default)
SIP/katrin-081b70a8200 (default)
SIP/katrin-081bb020200 (default)
SIP/hans-0816b8b8 150 (default)
which seems correct to me.
In case of a transfer of kwe to hans (katrin leaving), the result is:
pbxtest*CLI group show channels
ChannelGroup Category
SIP/kwe-081bf188 100 (default)
SIP/kwe-081bf188 200 (default)
SIP/hans-0816b8b8 150 (default)
I am confused about the second line, which leads to trouble. The above
context would think, that katrin is busy. In case of a blind transfer
everything is ok (the second line does not exist)
I have tested the above with * 1.4.14, 1.4.18-rc4 and 1.4.19
Is this a bug or a feature? Am I doing something wrong or should I file
a bug report?
Thanks in advance,
Regards
Karsten
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[Asterisk-Users] problem with sip registration with database
Hi all I have setup sips accounts to an asterisk server from a provider, I know that there are using asterisk real time for sip users definitions. Sometimes in a ramdom basis I receive: chan_sip.c:9596 handle_response_register: Forbidden - wrong password on authentication for REGISTER And i cannot make or receive calls. Nothing has change in configurations file, and if I make reload chan_sip.so, everything goes right again. I have set up in sip.conf following parameters: registertimeout=50 registerattempts=0 but it seems that there is no effect. Could it be due to database interacction?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with sip registration ramdomly
Hi all I have setup sips accounts to an asterisk server from a provider, I know that there are using asterisk real time for sip users definitions. Sometimes in a ramdom basis I receive: chan_sip.c:9596 handle_response_register: Forbidden - wrong password on authentication for REGISTER And i cannot make or receive calls. Nothing has change in configurations file, and if I make reload chan_sip.so, everything goes right again. I have set up in sip.conf following parameters: registertimeout=50 registerattempts=0 but it seems that there is no effect. Could it be due to database interacction. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP register
Hi! I'm registering an asterisk server in a Sysmaster with a SIP account. The registration succeeds and I can establish a call that come from the Sysmaster. After around 80 seconds the Sysmaster sends a BYE SIP message and the call hang up. This does not occur to the hard/soft SIP phones registered in the sysmaster. I debug, but the only info that I can get is the BYE message. Thanks for your suggetions soving the problem. Bye. -- Diego Andrés Asenjo González Universidad del Cauca Ingeniero en Electrónica y Telecomunicaciones signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP register
Diego Andrés Asenjo González wrote: Hi! I'm registering an asterisk server in a Sysmaster with a SIP account. The registration succeeds and I can establish a call that come from the Sysmaster. After around 80 seconds the Sysmaster sends a BYE SIP message and the call hang up. This does not occur to the hard/soft SIP phones registered in the sysmaster. I debug, but the only info that I can get is the BYE message. Thanks for your suggetions soving the problem. Bye. Hi, Enable SIP debug and check which peer sends BYE at first. After call establishment, can you hear voice for 80 sec.? What about RTP in this duration? -- Baris Simsek http://www.enderunix.org/simsek/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP channels
Hi all, i've a problem in my Asterisk system. We have around 30 SIP phones connected to an asterisk system, and sometimes some SIP channel (associated to an extension) gets busy all the time, even whenthat extensionisn't in use. We have a workaround for this, as we can't restart asterisk in work hours, we assign other extension to that phone and create an alias for calling there. When asterisk is restarted, all extensions aswer the way it's supposed to be. Any clues? Thank you :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP channels
Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas wrote: Hi all, i've a problem in my Asterisk system. We have around 30 SIP phones connected to an asterisk system, and sometimes some SIP channel (associated to an extension) gets busy all the time, even when that extension isn't in use. [..] Any clues? Without any debugging output from your Asterisk server guesses will range from bad SIP phones to bad Asterisk configuration or a small possibility of a bug. When reporting problems like this, you always have to mention which version of Asterisk you are using, which platform and which brand of phone. The more details you deliver, the more likely you will get a good answer that will help you forward. Without any details, you will only get bad answers or answers that will ask you more questions. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem setting SIP incoming/outgoing
Hi I am a newbie to * and I am having a problem which appears 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:[EMAIL PROTECTED]: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 fromdomain=sipserver.com host=sipserver.com port=8080 context=default with the above configuration I can successfully dial out using dial(SIP/[EMAIL PROTECTED]) but now when I call my incoming number, I get a busy or invalid number signal. If I coment out sipserverout section, I could receive incoming calls again. So I turned on sip debug on CLI. and it appears to me that the following is happening. astreisk takes the incoming call and tries to match it with a section with the same hostname. Now the reverse IP lookup on 109.147.41.48 return sipserver.com (which is correct), so it is trying to send the call to sipserverout which is essentially back to the same server where it came from (Notice the statement Found peer 'sipserverout' in the sip debug logs below). This creates an endless loop and the equipment at the other end terminates the call. According to all the examples I have seen, my setup is the correct setup and everyone seems to be using it. but it does not work for me. I am deperately looking for a solution. Please help. I am using asterisk 1.2.0 beta 1 on FC1. Here is the sip debug dump when a call is coming. -- SIP read from 109.147.41.48:8080: INVITE sip:[EMAIL PROTECTED]:5050 SIP/2.0 Record-Route: sip:209.47.41.48:80;ftag=2C996308-10F9;lr=on Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0 Via: SIP/2.0/UDP 209.47.41.61:5060;rport=53084;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4BB6EA6 From: sip:[EMAIL PROTECTED];tag=2C996308-10F9 To: sip:[EMAIL PROTECTED] Date: Thu, 06 Oct 2005 08:13:58 GMT Call-ID: [EMAIL PROTECTED] Supported: timer Min-SE: 1800 Cisco-Guid: 4208765565-896995802-2793406481-2459445924 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 4 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1128586438 Contact: sip:[EMAIL PROTECTED]:53084 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 369 hint: NAThelper hint: SDP rewritten hint: usrloc applied hint: NAT... v=0 o=CiscoSystemsSIP-GW-UserAgent 5168 3221 IN IP4 209.47.41.61 s=SIP Call c=IN IP4 109.147.41.48 t=0 0 m=audio 53870 RTP/AVP 0 8 18 3 101 c=IN IP4 109.147.41.48 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes --- (26 headers 16 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 109.147.41.48 : 80 (non-NAT) Found peer 'sipserverout' Reliably Transmitting (no NAT) to 209.47.41.48:80: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0 Via: SIP/2.0/UDP 209.47.41.61:5060;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4BB6EA6 From: sip:[EMAIL PROTECTED] ;tag=2C996308-10F9 To: sip:[EMAIL PROTECTED] ;tag=as1b7fff99 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED]:5050 Proxy-Authenticate: Digest realm=asterisk, nonce=6d00a83d Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from 109.147.41.48:8080: ACK sip:[EMAIL PROTECTED]:5050 SIP/2.0 Via: SIP/2.0/UDP 109.147.41.48:8080;branch=z9hG4bK03a4.da6a926.0 From: sip:[EMAIL PROTECTED];tag=2C996308-10F9 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as1b7fff99 CSeq: 101 ACK User-Agent: Phone Server 1 Content-Length: 0 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem setting SIP incoming/outgoing
Hi I am a newbie to * and I am having a problem which appears 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:[EMAIL PROTECTED]: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 fromdomain=sipserver.com host=sipserver.com port=8080 context=default with the above configuration I can successfully dial out using dial(SIP/[EMAIL PROTECTED]) but now when I call my incoming number, I get a busy or invalid number signal. If I coment out sipserverout section, I could receive incoming calls again. So I turned on sip debug on CLI. and it appears to me that the following is happening. astreisk takes the incoming call and tries to match it with a section with the same hostname. Now the reverse IP lookup on 109.147.41.48 return sipserver.com (which is correct), so it is trying to send the call to sipserverout which is essentially back to the same server where it came from (Notice the statement Found peer 'sipserverout' in the sip debug logs below). This creates an endless loop and the equipment at the other end terminates the call. According to all the examples I have seen, my setup is the correct setup and everyone seems to be using it. but it does not work for me. I am deperately looking for a solution. Please help. I am using asterisk 1.2.0 beta 1 on FC1. Here is the sip debug dump when a call is coming. -- SIP read from 109.147.41.48:8080: INVITE sip:[EMAIL PROTECTED]:5050 SIP/2.0 Record-Route: sip:209.47.41.48:80;ftag=2C996308-10F9;lr=on Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0 Via: SIP/2.0/UDP 209.47.41.61:5060;rport=53084;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4B B6EA6 From: sip:[EMAIL PROTECTED];tag=2C996308-10F9 To: sip:[EMAIL PROTECTED] Date: Thu, 06 Oct 2005 08:13:58 GMT Call-ID: [EMAIL PROTECTED] Supported: timer Min-SE: 1800 Cisco-Guid: 4208765565-896995802-2793406481-2459445924 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 4 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1128586438 Contact: sip:[EMAIL PROTECTED]:53084 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 369 hint: NAThelper hint: SDP rewritten hint: usrloc applied hint: NAT... v=0 o=CiscoSystemsSIP-GW-UserAgent 5168 3221 IN IP4 209.47.41.61 s=SIP Call c=IN IP4 109.147.41.48 t=0 0 m=audio 53870 RTP/AVP 0 8 18 3 101 c=IN IP4 109.147.41.48 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes --- (26 headers 16 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 109.147.41.48 : 80 (non-NAT) Found peer 'sipserverout' Reliably Transmitting (no NAT) to 209.47.41.48:80: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0 Via: SIP/2.0/UDP 209.47.41.61:5060;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4BB6EA6 From: sip:[EMAIL PROTECTED] ;tag=2C996308-10F9 To: sip:[EMAIL PROTECTED] ;tag=as1b7fff99 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED]:5050 Proxy-Authenticate: Digest realm=asterisk, nonce=6d00a83d Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from 109.147.41.48:8080: ACK sip:[EMAIL PROTECTED]:5050 SIP/2.0 Via: SIP/2.0/UDP 109.147.41.48:8080;branch=z9hG4bK03a4.da6a926.0 From: sip:[EMAIL PROTECTED];tag=2C996308-10F9 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as1b7fff99 CSeq: 101 ACK User-Agent: Phone Server 1 Content-Length: 0 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem setting SIP incoming/outgoing
I am a newbie to * and I am having a problem which appears 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:[EMAIL PROTECTED]: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
fromdomain=sipserver.com
host=sipserver.com
port=8080
context=default
with the above configuration I can successfully dial out using
dial(SIP/[EMAIL PROTECTED])
but now when I call my incoming number, I get a busy or invalid number
signal. If I coment out sipserverout section, I could receive incoming calls
again.
So I turned on sip debug on CLI. and it appears to me that the following is
happening. astreisk takes the incoming call and tries to match it with a
section with the same hostname. Now the reverse IP lookup on 109.147.41.48
return sipserver.com (which is correct), so it is trying to send the call to
sipserverout which is essentially back to the same server where it came from
(Notice the statement Found peer 'sipserverout' in the sip debug logs
below). This creates an endless loop and the equipment at the other end
terminates the call.
According to all the examples I have seen, my setup is the correct setup and
everyone seems to be using it. but it does not work for me. I am deperately
looking for a solution. Please help.
I am using asterisk 1.2.0 beta 1 on FC1.
In very general terms, you probably want something like this in your sip.conf:
[sipserver]
type=friend
secret=secret
username=user
fromuser=user
fromdomain=sipserver.com
host=sipserver.com
port=8080
insecure=very
canreinvite=no
dtmfmode=inband
context=from-sipserver
disallow=all
allow=ulaw
For sip stuff, notice the use of type=friend and canreinvite=no. The use
of the register statement (in this case) implies use of type=friend (for
both incoming and outgoing calls).
Then in extensions.conf, use something like this:
exten = _1NX,3,Dial(SIP/sipserver/${EXTEN})
where SIP/sipserver is referring to the context [sipserver] in sip.conf.
Did the folks at sipserver.com tell you to use port=8080? If not,
remove that statement as the default for sip is port=5060.
There are other ways to accomplish the same thing, so consider the above
as only way to do it.
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[Asterisk-Users] Problem: Got SIP response 481 Call Leg/Transaction Does Not Exist
I am not able to get softphone registered (active) with * . new installation , new user Able to get server started , and phone appears to register gets the SIP reponse 481 message Register SIP 4009 at 192.168.200.10 port 2199 expires 120 Unregistered SIP 4009 Register SIP 4009 at 192.168.200.10 port 9428 expires 120 Saved useragent RTC/1.24949 for peer 4009 Got SIP response 481 Call Leg/Transaction Does Not Exist back from 192.168.200.10 Got SIP response 481 Call Leg/Transaction Does Not Exist back from 192.168.200.10 NOTICE[19714]: chan_sip.c:9017 handle_reponse_peerpoke: Peer 4009 is now TOO LAGGED! (1780ms / 100ms) Got SIP response 481 Call/Leg .. System Info: Redhat 9.0 running on vmware Softphone: adoresoftphone If more information required to help resolve let me know thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem client sip (ser) to client sip (asterisk)
i am configure ser:
if (method==INVITE) {
if (uri=~sip:[EMAIL PROTECTED]) {
rewritehostport(192.168.0.183:5080);
};
};
an asterisk:
sip.conf
; config Xlite
[1234]
;context=sip
context=from-ser
type=friend
auth=md5
username=1234
secret=chooseapassword
;fromdomain=sorcier.com.pe ; para prueba de ser -asterisk
callerid=First Extension 1234
host=dynamic
canreinvite=no
;disallow=all
;allow=gsm
;allow=ulaw
;allow=alaw
;and conexion the ser to asterisk
;
[ser-sip]
type=friend; permitimos llamadas entrantes y
salientes. Usar peer si solo es MWI
context=ser-asterisk ; este es el contexto que usan las
llamadas entrantes
;host=sorcier.com.pe ; Este es tu hostname o IP del servidor SER
host=192.168.0.183
fromdomain=sorcier.com.pe ; este es tu SER_DOMAIN (nombre de dominio del SER)
;insecure=very ; Permite que las llamadas que viene del
SER pasen a Asterisk
insecure=yes
;[EMAIL PROTECTED] ; esto es para listar las cuentas de voicemail
;i am copy the voip-info
and the file the extensions.conf
; Configuracion al servidor ser, para llamada de ida
[from-ser]
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
[ser-asterisk]
; Ignora el dígito 0
;ignorepat = 0
; conexion a un telefono sip
;exten = _0X.,1,Dial(SIP/${EXTEN:1},90,Ttr)
;exten = _0X.,1,Dial(SIP/${EXTEN},20,Ttr)
;exten = _0X.,1,Dial(SIP/1234,20,Ttr)
;exten = _0X.,1,Dial(SIP/[EMAIL PROTECTED],20,Ttr)
exten = _0X.,1,Dial(SIP/${EXTEN})
i am probe diferents combinations, but no work
debug with asterisk and view itis:
Sip read:
INVITE sip:[EMAIL PROTECTED]:5080 SIP/2.0
Record-Route: sip:192.168.0.183;ftag=78607191;lr=on
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar sip:[EMAIL PROTECTED];tag=78607191
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 3143 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299
v=0
o=rbolivar 23151399 23151750 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.183 : 5060 (non-NAT)
Found peer 'ser-sip'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.185:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 1234 in ser-asterisk
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar sip:[EMAIL PROTECTED];tag=78607191
To: sip:[EMAIL PROTECTED];tag=as1ca211c4
Call-ID: [EMAIL PROTECTED]
CSeq: 3143 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]:5080
Content-Length: 0
to 192.168.0.183:5060
Sip read:
INVITE sip:[EMAIL PROTECTED]:5080 SIP/2.0
Record-Route: sip:192.168.0.183;ftag=78607191;lr=on
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar sip:[EMAIL PROTECTED];tag=78607191
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
CSeq: 3143 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299
v=0
o=rbolivar 23151399 23151750 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 13 lines
Ignoring this request
Sip read:
ACK sip:[EMAIL PROTECTED]:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0
From: rbolivar sip:[EMAIL PROTECTED];tag=78607191
Call-ID: [EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];tag=as1ca211c4
CSeq: 3143 ACK
User-Agent: Sip EXpress router(0.10.99-dev14-tcp (i386/linux))
Content-Length: 0
8 headers, 0 lines
Destroying call '[EMAIL PROTECTED]'
Sip read:
INVITE sip:[EMAIL PROTECTED]:5080 SIP/2.0
Record-Route: sip:192.168.0.183;ftag=78607191;lr=on
Via: SIP/2.0/UDP
[Asterisk-Users] problem calling SIP accounts
Hi I have configured sip accounts and they work some times. when i make a call to another SIP account it works right but some times i get the following error Jul 29 07:17:00 WARNING[802]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) this happence when i register the SIP users and stay for some time and dial.but no problem with out going calls, can call any time. Regards Kanishka ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP
Hello My name is Will. I have a problem with SIP on ASTERISK How many ways it has to register and to work in sip.conf? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem accepting sip call cvs head
Dear all, just upgraded to cvs head june 6th, using 1.0.7 sip.conf but can't accept any calls from SIP proxy. Anyone encountered the same problem? [general] context=sip-in recordhistory=yes ; Record SIP history by default port=5070 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=xxx.xxx.xxx.xxx; IP address to bind to asterisks(0.0.0.0 binds to all) rtptimeout=60 rtpholdtimeout=300 videosupport=yes tos=lowdelay tos=184 useragent=B2BUA canreinvite=no trustrpid=yes allowguest=yes trustrpid=yes autocreatepeer=yes disallow=all allow=g723 allow=g723.1 allow=g729 allow=gsm allow=alaw ; Allow codecs in order of preference allow=ulaw ; Note: codec order is respected only Ray ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP clients
Hi, I have my asterisk server inside a NAT. When i connect a softphone SIP inside my net, all go well. Ok. But when I try connect a SIP softphone client outside my NAT, I get: NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex sip:[EMAIL PROTECTED]' failed for '83.41.119.25' Can someone help me with this? PD: Sorry for my english ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with SIP clients
Are you doing port forwarding on your firewall? Just make sure your asterisk port is open... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Piqueras Sent: 30 May 2005 10:49 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with SIP clients Hi, I have my asterisk server inside a NAT. When i connect a softphone SIP inside my net, all go well. Ok. But when I try connect a SIP softphone client outside my NAT, I get: NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex sip:[EMAIL PROTECTED]' failed for '83.41.119.25' Can someone help me with this? PD: Sorry for my english ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP clients
Has you redirected all the RTP ports? You must redirect the SIP and the RTP streams. Take a look to the rtp.conf file of your asterisk installation to configure the RTP ports that you want to use. Best regards. Rpr Alex Piqueras escribió: Hi, I have my asterisk server inside a NAT. When i connect a softphone SIP inside my net, all go well. Ok. But when I try connect a SIP softphone client outside my NAT, I get: NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex sip:[EMAIL PROTECTED]' failed for '83.41.119.25' Can someone help me with this? PD: Sorry for my english ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP peer registration
Hi I am trying to get 2 incoming SIP accounts working from 2 different providers. One is sipgate.co.uk and the other is voipuser.org. If I load the Register command seperate they will both register phone and incoming works. If I try to load them both only sipgate registers. Anybody got any suggestions why? Regards Jon ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem in SIP md5 REGISTER
I guess I found a bug in the register logic in chan_sip
I'm trying of registering two extensions from a SIP gateway into Asterisk.
I have defined two user entries in sip.conf as follows:
[0191]
type = friend
auth=md5
username=0191
secret=planet
disallow=all
allow=ulaw
dtmfmode=inband
host = dynamic
default = 192.168.2.183
[0192]
type = friend
auth=md5
username=0192
secret=planet
disallow=all
allow=ulaw
dtmfmode=inband
host = dynamic
default = 192.168.2.183
And configured the gateway to register to asterisk (192.168.2.175) both
numbers with these username and passwords.
***
reg_num: 0191
Registrar_ID 1: UnRegistered
registrar: 192.168.2.175 5060expires: 600
name: 0191passwd: planet
reg_num: 0192
Registrar_ID 2: Registered
registrar: 192.168.2.175 5060expires: 600
name: 0192passwd: planet
***
When I reset the gateway I see the first sip user (0191) FAILS to
register, but the second one (0192) registers OK.
I first thought there was a problem with the digest response from the
gateway but after logging the SIP headers, and
reading the RFC's and use md5sum to check the digest values I realiced
the values from the cliente where OK.
In inserted some ast_log(LOG_NOTICE, ..) into the chan_sip.c 's
register_verify() and check_auth() functions
and found the problem is in Asterisk.
As you can see It seems for some reason when Asterisk receives both
REGISTER request messages one after the other,
he is mixing the nonce value (called randdata into chan_sip.c) for one
peer with the other.
So he ends evaluating the digest for the first register (0191) using the
nonce value from the second one (0192) and It fails.
For some reason (I think It is because the randdata is resetted to ''
after 0191 fails) the second register (0192) gets a second 407 Proxy
Authentication Required with a third randdata and this time It is
registered OK because the right nonce value is used.
I'm using Asterisk CVS version from 2004/05/19.
Here follow the console log (with my LOG_NOTICE debug messages) and the
corresponding ngrep SIP capture. Look specially the randdata values used
in check_auth (nonce value) and the (not) corresponding values sent in
the SIP responses for each REGISTER.
Everyone can check the response=... sent by the gateway are ok using
something like this:
A1=$(echo -n '0192:asterisk:planet'|md5sum|awk '{print $1}')
A2=$(echo -n 'REGISTER:sip:192.168.2.175'|md5sum|awk '{print $1}')
NONCE=17e63cd4
$(echo -n $A1:$NONCE:$A2|md5sum|awk '{print $1}')
**
*
Asterisk Console Logs
*
May 26 16:56:47 NOTICE[196621]: chan_sip.c:3861 register_verify:
Checking Auth: randata= name=0191 secret=planet uri=sip:192.168.2.175
May 26 16:56:47 NOTICE[196621]: chan_sip.c:3861 register_verify:
Checking Auth: randata=17e63cd4 name=0192 secret=planet
uri=sip:192.168.2.175
May 26 16:56:47 NOTICE[196621]: chan_sip.c:3861 register_verify:
Checking Auth: randata=49760cde name=0191 secret=planet
uri=sip:192.168.2.175
May 26 16:56:47 WARNING[196621]: chan_sip.c:3764 check_auth:
A1='0191:asterisk:planet'
May 26 16:56:47 WARNING[196621]: chan_sip.c:3769 check_auth:
resp_uri='sip:192.168.2.175' uri='sip:192.168.2.175'
May 26 16:56:47 WARNING[196621]: chan_sip.c:3770 check_auth:
A2='REGISTER:sip:192.168.2.175'
May 26 16:56:47 WARNING[196621]: chan_sip.c:3778 check_auth:
resp='160723a2f5a8dcf360271903c6818b63:49760cde:c70c5186f40f678679f57680d2a4390d'
resp_hash='267b05f67388676fcffb6bd3ee381b2e'
May 26 16:56:47 WARNING[196621]: chan_sip.c:3781 check_auth: Client
response='406d89d8d15ba1c9753b5bef95931934'
May 26 16:56:47 NOTICE[196621]: chan_sip.c:5691 handle_request:
Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.2.183'
May 26 16:56:48 NOTICE[196621]: chan_sip.c:3861 register_verify:
Checking Auth: randata= name=0192 secret=planet uri=sip:192.168.2.175
May 26 16:56:48 NOTICE[196621]: chan_sip.c:3861 register_verify:
Checking Auth: randata=23b5124b name=0192 secret=planet
uri=sip:192.168.2.175
May 26 16:56:48 WARNING[196621]: chan_sip.c:3764 check_auth:
A1='0192:asterisk:planet'
May 26 16:56:48 WARNING[196621]: chan_sip.c:3769 check_auth:
resp_uri='sip:192.168.2.175' uri='sip:192.168.2.175'
May 26 16:56:48 WARNING[196621]: chan_sip.c:3770 check_auth:
A2='REGISTER:sip:192.168.2.175'
May 26 16:56:48 WARNING[196621]: chan_sip.c:3778 check_auth:
resp='c04abf6412f4f786ba81daddb46a82ee:23b5124b:c70c5186f40f678679f57680d2a4390d'
resp_hash='c370755ec882aafa390ff867d1a99449'
May 26 16:56:48 WARNING[196621]: chan_sip.c:3781 check_auth: Client
response='c370755ec882aafa390ff867d1a99449'
interface: eth0 (192.168.2.0/255.255.255.0)
filter: ip and ( port 5060 and host
Re: [Asterisk-Users] Problem in SIP md5 REGISTER
Luis, I tried to simulate your situation using a sip agent (Xten X-Pro) and having it register to Asterisk with two user ids simultaneously all on the same LAN. I cannot replicate your problem. Both id's registered immediately. Can you test this in your environment replacing the gateway with another agent capable of dual proxy configuration? Also, in your friend definitions below: the correct parameter is defaultip and not default the auth option has been eliminated since it was never used for anything. Luis Vazquez wrote: I guess I found a bug in the register logic in chan_sip I'm trying of registering two extensions from a SIP gateway into Asterisk. I have defined two user entries in sip.conf as follows: [0191] type = friend auth=md5 username=0191 secret=planet disallow=all allow=ulaw dtmfmode=inband host = dynamic default = 192.168.2.183 [0192] type = friend auth=md5 username=0192 secret=planet disallow=all allow=ulaw dtmfmode=inband host = dynamic default = 192.168.2.183 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP softphone
Hi! On my SIP softphone, when I stop speaking the audio stops. So if im not talking I cant hear the other person. FAQ! X-Lite: Menu -- Advanced settings -- Audio -- Silence Set Transmit Silence to YES P. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP softphone
Hi! X-Lite: Menu -- Advanced settings -- Audio -- Silence set keep transmitting after silence to 1 or something like that Cf - Original Message - From: Philipp von Klitzing [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 21, 2004 11:24 AM Subject: Re: [Asterisk-Users] Problem with SIP softphone Hi! On my SIP softphone, when I stop speaking the audio stops. So if im not talking I cant hear the other person. FAQ! X-Lite: Menu -- Advanced settings -- Audio -- Silence Set Transmit Silence to YES P. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP softphone
Ok that fixed it. But why all of a sudden did it start doing this after I updated? Anyidea? It had been working fine for a few months. Kyle Philipp von Klitzing wrote: Hi! On my SIP softphone, when I stop speaking the audio stops. So if im not talking I cant hear the other person. FAQ! X-Lite: Menu -- Advanced settings -- Audio -- Silence Set Transmit Silence to YES P. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP softphone
Having a weird problem after I updated the other day. On my SIP softphone, when I stop speaking the audio stops. So if im not talking I cant hear the other person. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP softphone
On Thu, 2004-05-20 at 18:47, Kyle Hagan wrote: On my SIP softphone, when I stop speaking the audio stops. So if im not talking I cant hear the other person. http://lists.digium.com/pipermail/asterisk-users/2003-November/027732.html http://lists.digium.com/pipermail/asterisk-users/2003-November/027739.html http://lists.digium.com/pipermail/asterisk-users/2003-November/027806.html http://lists.digium.com/pipermail/asterisk-users/2004-February/035638.html -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problem with SIP configuration AND EXTENSION.
Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.6' Are you sure your phone isn't registering? These errors aren't related to your grandstream. Do a sip show peers at the Asterisk CLI and see if it shows your phone registered. I have in sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to register = [EMAIL PROTECTED]/phone ; 192.168.0.6 it´s my server linux ASTERISK. Take this line out. You don't need it. That's only for remote SIP providers. You're telling your * box to register with itself. And obviously bad things are happening! Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP 407
Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info (SP5002/S) and traed to register to asterisk, It seems to autentícate but sniffing the net it shows a 407 proxy authen required error message and I cannot make any outgoing calls from that gateway. Thats what I have: Sip.conf == [SOMESIP] type=friend secret=xxx insecure=no ;host=192.168.6.2 host=dynamic defaultip=192.168.6.2 context=nacer mailbox=601 On the FXS: usr/config$ sip -print Run Mode : PROXY MODE Proxy address: 192.168.2.2 Proxy port : 5060 Domain : null Prefix string: null Line1: 1001 Line2: 1002 SIP listen port : 5060 RTP receive port : 16384 Expire : 3600 usr/config$ security -print Line1 account information Username: SOMESIP Password: x Ive tried putting Domain = Asterisk on the FXS and other things, also played with codecs but everything seems to come from the 407 message, how can I avoid that message? Another thing is that I need to register the gateway, so it doesnt allow calling out if it hasnt registered. Thanks! Marc! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP 407
G'day Marc, On Wed, 25 Feb 2004, Marc Fargas wrote: Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info (SP5002/S) and traed to register to asterisk, It seems to autentícate but sniffing the net it shows a 407 proxy authen required error message and I cannot make any outgoing calls from that gateway. I captured a flow between * and an ATA-186 the other day, because I had the same problem (well, the symptom was the same). The 407 message from * is part of the registration flow. It tells the client that it needs to resend its REGISTER, this time including a Proxy-Authentication (sp?) header in the request. That header contains the authentication data (authuser, password). I'd suggest getting into the network with ethereal or the like and start sniffing the packet flow. In my case, a hardware incompatibility was preventing my client from receiving the 407 from *, so it never responded to it... (Getting a packet trace will also be essential in getting further support, either from your FXS vendor or the SIP mavens on this list.) Ive tried putting Domain = Asterisk on the FXS and other things, also played with codecs but everything seems to come from the 407 message, how can I avoid that message? Well, you could try removing the password (secret=XXX) from the entry in sip.conf, allowing the client to register without authentication. Might be something to try, but I don't think I'd run live that way... ;-) Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP 407
Vic Cross wrote: G'day Marc, On Wed, 25 Feb 2004, Marc Fargas wrote: Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info (SP5002/S) and traed to register to asterisk, It seems to autentcate but sniffing the net it shows a 407 proxy authen required error message and I cannot make any outgoing calls from that gateway. I captured a flow between * and an ATA-186 the other day, because I had the same problem (well, the symptom was the same). The 407 message from * is part of the registration flow. It tells the client that it needs to resend its REGISTER, this time including a Proxy-Authentication (sp?) header in the request. That header contains the authentication data (authuser, password). Let's clear this up: A SIP ua sends a REGISTER to a location server to tell the server where it can be reached. At registration, the server challenges the UA with a www-authentication. When authenticated, the server stores the IP address and contact header for some time (expiry=) to be able to place calls to the UA. This is a SIP peer in asterisk. The standard sip channels has a bug here and issues a Proxy-authentication. The chan_sip2 channel issues a www-auth. When a SIP UA want to call through asterisk, asterisk want's to know for certain who it is before admitting any services (except default context). To let the SIP ua through, we issue a Proxy-auth. If it succeeds, the asterisk sip user is allowed to reach whatever is reachable in the user's SIP context. A type=friend SIP client is both a user and a peer. Neither form of authentication sends the password in clear. This is nowadays forbidden in SIP. We use digest authentication, a challenge-response mechanism. I'm a bit afraid that Asterisk's authentication in the SIP channels is a bit out of date and that may be your problem. Please forward SIP debug output so we can go through the various stages that leads to the 407. Ive tried putting Domain = Asterisk on the FXS and other things, also played with codecs but everything seems to come from the 407 message, how can I avoid that message? Well, you could try removing the password (secret=XXX) from the entry in sip.conf, allowing the client to register without authentication. Might be something to try, but I don't think I'd run live that way... ;-) If so, add an ACL so you limit the IP addresses that may use this account. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with SIP 407
On this cuts note that the gateway has username 'Republica', you could see some reference to Republica2 which corresponds to a second line on the gateway that I have disabled. Thanks for your help! That's SIP debug when dialling '9' (9 would do Goto(s,1)) === *CLI *CLI 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK0f92815a From: asterisk sip:[EMAIL PROTECTED];tag=as0bc66d50 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.6.2:5060 Sip read: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.2.2:5060;received=192.168.2.2;branch=z9hG4bK0f92815a From: asterisksip:[EMAIL PROTECTED] ;tag=as0bc66d50 To: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30d-bef46-6225 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Content-Length:0 7 headers, 0 lines Feb 25 21:03:04 WARNING[98311]: chan_sip.c:4875 handle_response: Host '192.168.6.2' does not implement 'NOTIFY' Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.6.2:5060 Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: FXS_GW (2asipfxs.106) From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Content-Type: application/sdp Content-Length:234 v=0 o=FXS_GW 12367 0 IN IP4 192.168.6.2 s=Audio Session i=Audio Session c=IN IP4 192.168.6.2 t=0 0 m=audio 16384 RTP/AVP 18 4 0 8 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 10 headers, 11 lines Using latest request as basis request Sending to 192.168.6.2 : 5060 (non-NAT) Found audio format UNKN Found audio format ULAW Found audio format UNKN Found audio format ALAW Found description format G729 Found description format G723 Found description format PCMU Found description format PCMA Capabilities: us - 6, them - 269/854015, combined - 6 Non-codec capabilities: us - 1, them - 0, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.6.2:5060 From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as76b77fc8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=2f9d85fe Content-Length: 0 to 192.168.6.2:5060 Sip read: ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.6.2:5060 Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: FXS_GW (2asipfxs.106) From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone ;tag=as76b77fc8 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Content-Length:0 9 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.6.2:5060 Contact: sip:[EMAIL PROTECTED]:5060 Proxy-Authorization: Digest username=Republica, realm=asterisk, nonce=2f9d85fe, uri=sip:[EMAIL PROTECTED]:5060;user=phone, re sponse=4b434a0e18166c573b006cf9cbd2f3bc, algorithm=MD5 User-Agent: FXS_GW (2asipfxs.106) From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Content-Type: application/sdp Content-Length:234 v=0 o=FXS_GW 12367 0 IN IP4 192.168.6.2 s=Audio Session i=Audio Session c=IN IP4 192.168.6.2 t=0 0 m=audio 16384 RTP/AVP 18 4 0 8 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 11 headers, 11 lines Using latest request as basis request Sending to 192.168.6.2 : 5060 (non-NAT) Found audio format UNKN Found audio format ULAW Found audio format UNKN Found audio format ALAW Found description format G729 Found description format G723 Found description format PCMU Found description format PCMA Capabilities: us - 6, them - 269/854015, combined - 6 Non-codec capabilities: us - 1, them - 0, combined - 1 Looking for 9 in nacer list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.6.2:5060 From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-30e-befaf-7ba1 To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=as76915db6 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.6.2:5060 -- Executing Goto(SIP/Republica-a2aa, default|s|1) in new stack -- Goto (default,s,1) -- Executing Answer(SIP/Republica-a2aa, ) in new stack We're at 192.168.2.2 port 19466 Video is at 192.168.2.2 port 18490 Answering with preferred capability 4 Answering with preferred capability 2 Answering with
Re: [Asterisk-Users] Problem with SIP 407
Seems like republica registers ok, but not republica2. Republica2 failes to authenticate. You have a normal registration sequense here: -Client sends a REGISTER without authentication -Server sends trying... -Server sends 407 Proxy auth (should be WWW auth) with challenge -Clients ACK -Client sends a new REGISTER with authentication -Server tries auth -If auth fails (propably wrong secret/password) a 401 unauthorized is issued -If auth succeeds a 200 OK is issued Apart from that, the Asterisk server after authentication wants to tell the client that it has voicemail, and the client responds that it has no clue of what the server is trying to say. Take away the mailbox= parameter in sip.conf to avoid this. You client seems to send a lot of REGISTERs without waiting for response. Other than that, check the passwords for republica2. Republica1 should be able to receive calls from asterisk. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with SIP 407
I have commented de Republica2 in sip.conf 'cause if I uncomment it neither Republica and Republica2 register (maybe because they're on the same gateway?) Well, inspite it register well when I try tocall any extension It plays 'busy' tone immediately after Asterisk takes the calls I thought it was a codecs problem but I have another gateway similar with H.323 and hav codecs configured same way both on asterisk and the gateways, the H.323 one goes right but the SIP one can't do anything, it just plays around with 'busy' tones. In my previous post you can see the output of sip debug on Asterisk when trying to call an extension, on the gateway side that's what I get: Line : 1, Start Inviting strDes To:sip:[EMAIL PROTECTED]:5060;user=phone, strOri From:sip:[EMAIL PROTECTED] 1-RvSipCallLegMgrCreateCallLeg() ok! Success to rvSdpMsgEncodeToBuf * -- Message Sent (Message type: 0) (call-leg 58e04c) INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.6.2:5060 Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: FXS_GW (2asipfxs.106) From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-403d1740-5870d2-7301 To: sip:[EMAIL PROTECTED]:5060;user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Content-Type: application/sdp Content-Length:234 v=0 o=FXS_GW 12367 0 IN IP4 192.168.6.2 s=Audio Session i=Audio Session c=IN IP4 192.168.6.2 t=0 0 m=audio 16384 RTP/AVP 18 4 0 8 a=rtpmap:18 G729/8000/1 a=rtpmap:4 G723/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 1-RVSIP_CALL_LEG_STATE_INVITING -- Message Received (Message Type: 1) (call-leg 58e04c) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.6.2:5060 User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER Contact: sip:[EMAIL PROTECTED] From: sip:[EMAIL PROTECTED] ;tag=c0a80602-13c4-403d1740-5870d2-7301 To: sip:[EMAIL PROTECTED]:5060;user=phone ;tag=as07a0b938 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Content-Length:0 1-RVSIP_CALL_LEG_STATE_TERMINATED 1-Gen_BusyTone ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP-Phones and * audio-files
On Fri, 2003-11-28 at 15:26, Ernst Lehmann wrote: Hi All, I am a newbie to asterisk, and here is my first problem, where I do not know any further. I have to grandstream BT100 connected to asterisk. Working fine, for calling to each other, and to call via a IAX-Link to the outside. If I try to call the initial demo from the samples.extensions.conf I have nothing to hear. I solved the problem. Just for the archives :-)) It was the not connected E100P Card, because of this, there was no timing-device I think. After unloading the modules for the e100p card, and loading the zaprtc module. It worked, without any problem. [] -- Bye Ernst - Ernst Lehmann Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP-Phones and * audio-files
Hi All, I am a newbie to asterisk, and here is my first problem, where I do not know any further. I have to grandstream BT100 connected to asterisk. Working fine, for calling to each other, and to call via a IAX-Link to the outside. If I try to call the initial demo from the samples.extensions.conf I have nothing to hear. The CLI fine reports: -- Executing Playback(SIP/2209-0260, demo-abouttotry) in new stack -- Playing 'demo-abouttotry' (language 'en') after a few seconds, when I give it up == Spawn extension (demo, 500, 1) exited non-zero on 'SIP/2209-0260' When I call to the voicemail-system with extension 8500, I got also only silence on the phone. What can it bee ?? I tried asterisk with cvs from today (28-11-2003) and with an older version cvs from (19-11-2003) Thanks for any hints something about the hardware: - P4 2.8 GHz - 1 GB RAM - Digium E100P (but not connected at the moment) - Digium TDM400P (but also not connected to devices at the moment) -- Here my additions to the sip.conf disallow=all allow=ulaw allow=alaw allow=g729 allow=g723.1 allow=gsm allow=ilbc allow=speex allow=lpc10 ; my grandstream 102 [2209] type=friend username=2209 secret=nosecretpasswordhere host=dynamic context=demo canreinvite=yes dtmfmode=info qualify=yes disallow=all allow=g723.1 allow=ulaw allow=alaw allow=gsm ; my grandstream 102 [2210] type=friend username=2210 secret=nosecret host=dynamic context=demo canreinvite=yes dtmfmode=info qualify=yes disallow=all allow=ulaw allow=gsm allow=alaw -- in extensions.conf I only added this to lines under section [demo] for testing the calls from gs1 - gs2 exten = 2209,1,Dial(SIP/2209) exten = 2210,1,Dial(SIP/2210) - -- Bye Ernst - Ernst Lehmann Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP and DOS attacks...
There was a tread that I googled for and could not find about Asterisk being open to SIP DOS Attacks. I have a customer whose machine was hammered last light by traffic on its SIP port causing the OS to use up its resources. Namely number of open files. The discussion was around the fact that the Sip protocol answers requests without regard to authentication. Can anyone comment on this
Re: [Asterisk-Users] Problem with SIP and DOS attacks...
On Wed, 2003-10-15 at 15:22, Alex Lopez wrote: There was a tread that I googled for and could not find about Asterisk being open to SIP DOS Attacks. I have a customer whose machine was hammered last light by traffic on its SIP port causing the OS to use up its resources. Namely number of open files. The discussion was around the fact that the Sip protocol answers requests without regard to authentication. Can anyone comment on this You had limited google help due to your misunderstanding of the problem. Use asterisk sip vulnerability http://www.google.com/search?hl=enie=UTF-8oe=UTF-8q=asterisk+sip+vulnerabilitybtnG=Google+Search This is not a DoS, it is a remote exploit. Since you seemed to not understand it by the above message I'll give a quick run down of the two different types of attack. A DoS attack can be as simple as a flood of messages. It could be specially crafted messages that require your computer to bog down trying to service them, or just a large number of them. A remote exploit means that you can run certain code from remote without authentication. As in most of us run asterisk as root, so anyone that is able to instruct asterisk to do something will get it run by the root user. Next, if you had been a competent admin, you would have done your updates on all the machines back then since the update was put into CVS around 8-15. If you are 2 months behind on your patching, you need to consider tools that help you get this done. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP authentication
It looks like you are registering fine. If you dial 12321 from another phone, does it not ring? This is the transaction as I see it in the log that you attached: Phone: REGISTER Asterisk: Proxy Authentication Required (Send me your credentials) Phone: REGISTER with CREDENTIALS Asterisk: 200 OK (You are now registered) Asterisk: NOTIFY (You have 0/0 messages in your voicemail.) Phone: 200 OK (Thanks for letting me know) Sean ___ Sean Robertson NETXUSAp. 800-289-6389f. 864-233-4344 "Ask me about Voice over IP."http://www.netxusa.com/ - Original Message - From: John Foster To: [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 12:49 AM Subject: [Asterisk-Users] Problem with SIP authentication Hi List, After going through mailing list and manual of asterisk, I still could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params [12321]type=friendusername=12321host=dynamicsecret=ccartacontext=defaultmailbox=1234,2345 ; Mailbox for message waiting indicator [7]type=friendusername=7host=dynamicsecret=atracccontext=defaultmailbox=1234,2345 m trying with SIPPS UA, that gets status "Acquired", not "Registered", Can anyone give any idea about it? I tried same with X-Lite, didnt work. Sip debug messages are pasted below. Best Regards, JF Sip read:REGISTER sip:192.168.100.71 SIP/2.0Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2cf0baTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]Contact: ccarta sip:[EMAIL PROTECTED]:5062;expires=600;q=0.500Expires: 600CSeq: 1 REGISTERContent-Length: 0User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 10 headers, 0 linesUsing latest request as basis requestSending to 192.168.100.66 : 5062 (non-NAT)Transmitting (no NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2cf0baTo: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 0 to 192.168.100.66:5062Transmitting (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2cf0baTo: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Proxy-Authenticate: Digest realm="asterisk", nonce="7c7fba4d"Content-Length: 0 to 192.168.100.66:5062Sip read:REGISTER sip:192.168.100.71 SIP/2.0Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2d0018To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]Contact: ccarta sip:[EMAIL PROTECTED]:5062;expires=600;q=0.500Expires: 600CSeq: 2 REGISTERContent-Length: 0Proxy-Authorization: Digest username="12321",realm="asterisk",uri="sip:192.168.100.66",nonce="7c7fba4d",nc="0001",response="b8b1d7fc53eff354dfc31dfa3f800749"User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 11 headers, 0 linesUsing latest request as basis requestSending to 192.168.100.66 : 5062 (non-NAT)Transmitting (no NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2d0018To: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 0 to 192.168.100.66:5062Transmitting (no NAT):SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2d0018To: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERExpires: 600Contact: sip:[EMAIL PROTECTED];expires=600Date: Tue, 14 Oct 2003 13:46:14 GMTContent-Length: 0 to 192.168.100.66:506211 headers, 2 linesReliably Transmitting:NOTIFY sip:[EMAIL PROTECTED]:5062 SIP/2.0Via: SIP/2.0/UDP 192.168.100.71:5060;branch=z9hG4bK3ecb7a3bFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as3f6e8c0eTo: sip:[EMAIL PROTECTED]:5062Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: Asterisk PBXEvent: message-summaryContent-Type: application/simple-message-summa
[Asterisk-Users] Problem with SIP authentication
Hi List, After going through mailing list and manual of asterisk, I still could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params [12321]type=friendusername=12321host=dynamicsecret=ccartacontext=defaultmailbox=1234,2345 ; Mailbox for message waiting indicator [7]type=friendusername=7host=dynamicsecret=atracccontext=defaultmailbox=1234,2345 m trying with SIPPS UA, that gets status "Acquired", not "Registered", Can anyone give any idea about it? I tried same with X-Lite, didnt work. Sip debug messages are pasted below. Best Regards, JF Sip read:REGISTER sip:192.168.100.71 SIP/2.0Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2cf0baTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]Contact: ccarta sip:[EMAIL PROTECTED]:5062;expires=600;q=0.500Expires: 600CSeq: 1 REGISTERContent-Length: 0User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 10 headers, 0 linesUsing latest request as basis requestSending to 192.168.100.66 : 5062 (non-NAT)Transmitting (no NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2cf0baTo: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 0 to 192.168.100.66:5062Transmitting (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2cf0baTo: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Proxy-Authenticate: Digest realm="asterisk", nonce="7c7fba4d"Content-Length: 0 to 192.168.100.66:5062Sip read:REGISTER sip:192.168.100.71 SIP/2.0Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2d0018To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]Contact: ccarta sip:[EMAIL PROTECTED]:5062;expires=600;q=0.500Expires: 600CSeq: 2 REGISTERContent-Length: 0Proxy-Authorization: Digest username="12321",realm="asterisk",uri="sip:192.168.100.66",nonce="7c7fba4d",nc="0001",response="b8b1d7fc53eff354dfc31dfa3f800749"User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 11 headers, 0 linesUsing latest request as basis requestSending to 192.168.100.66 : 5062 (non-NAT)Transmitting (no NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2d0018To: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 0 to 192.168.100.66:5062Transmitting (no NAT):SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66From: sip:[EMAIL PROTECTED];tag=3b2d0018To: sip:[EMAIL PROTECTED];tag=as648287faCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERExpires: 600Contact: sip:[EMAIL PROTECTED];expires=600Date: Tue, 14 Oct 2003 13:46:14 GMTContent-Length: 0 to 192.168.100.66:506211 headers, 2 linesReliably Transmitting:NOTIFY sip:[EMAIL PROTECTED]:5062 SIP/2.0Via: SIP/2.0/UDP 192.168.100.71:5060;branch=z9hG4bK3ecb7a3bFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as3f6e8c0eTo: sip:[EMAIL PROTECTED]:5062Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: Asterisk PBXEvent: message-summaryContent-Type: application/simple-message-summaryContent-Length: 36 Messages-Waiting: noVoicemail: 0/0(no NAT) to 192.168.100.66:5062Sip read:SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.100.71;branch=z9hG4bK3ecb7a3bFrom: "asterisk" sip:[EMAIL PROTECTED];tag=as3f6e8c0eTo: sip:[EMAIL PROTECTED]:5062;tag=3b302259Call-ID: [EMAIL PROTECTED]CSeq: 102 NOTIFYUser-Agent: Ahead SIPPS IP Phone Version 2.0.42.13Content-Length: 0 8 headers, 0 lines Do you Yahoo!? The New Yahoo! Shopping - with improved product search
[Asterisk-Users] Problem with SIP Client!
Ok I have the following on the Asterisk every minutes. Got SIP response 481 Call Leg/Transaction Does Not Exist back from 2XX.2XX.133.1XX. The user of this Sip phone is able to make calls and get calls. A Windows 2000 pro using MS Messenger! I loaded it on my PC as well and it does the same for my IP address! Is there some thing I need change on it! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP: Maximum retries exceeded
Hi all, this message occurs if i was connected or not: WARNING[213006]: File chan_sip.c, Line 432 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Response) If i was connected, the call will be disconnected after a few seconds. What does it means ? I don't see anything to configure like Max retries Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP Native Bridging and UPnP
My configuration is comprised of two Snom 200 phones, two FXO cards connected to two PSTN lines, and one SIP account at iConnect. Snom1 has a VPN connection to the remote Asterisk server. Snom2 is using UPnP behind a Linksys WRT54G router/firewall to connect to the same server. All outgoing calls are routed to iConnect. Snom1 works correctly. Snom2 can call Snom1, and receive calls from the PSTN, but calls made to the PSTN through iConnect complete but nothing can be heard. The debug output shows that Asterisk is attempting a native bridge between iConnect and Snom2 at the time the call is answered and the line goes dead. It would seem that a native bridge is not working between iConnect and the phone behind a UPnP firewall. Is the UPnP device the problem? Can it be configured to fix the problem? If not, can native bridging be disabled for this phone? Otherwise, it would seem that I will have to establish a VPN to this phone just to solve this problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP Phone with outgoing phone call
I have a X100P and am calling out from a desktop within the same network. I connect to * then dialout a local phone number to my cell phone. It rings 2 times then hangs up. I'mtesting Sipps as the softphone. * is saying "retries exceeded". Has anyone had this problem? It's probably with my sip.conf. John
