Pickup detection problem

Asterisk Asterisk Support
1

Hi,
I am running Asterisk 1.6.2.13
I have a problem on one phone (only), which does not detect a call pickup, this phone is a Nortel 1535.
I can give a ring without any problem to this phone, but when I call a number from this phone, it does not detect that the other side picks up the phone, and it keeps ringing.
Weird thing is that I can see RTP traffic between the 2 phones, though the Nortel is still ringing. I have tried several parameters without any success :

dtmfmode = rfc2833
rfc2833compensate = yes
progressinband = yes

Any idea ?
Thanks very for your help as I run out of ideas.

Fabien

2

just on thing that might help troubleshooting this problem :

If I give a ring from the Nortel to a phone A, when A picks-up , I can see “call connected” on the Nortel, but the Nortel is still ringing. Then if on the Nortel, I put the communication on-hold, and resume it, it stops ringing and I can talk to phone A and hear it as well…that’s crazy !

Cheers,
Fabien

3

Fabian,

I think that the problem could be on the SIP messages.

Could you please enable SIP DEBUG in asterisk and check if Asterisk is sending the “200 OK” to the Nortel when the other phones answers the call.

4

Hi,
Yes apparently I have a SIP 200 OK :

192.168.1.18 is the Nortel Phone form which I give the call with extension 121.
192.168.1.72 is the phone receiving the call with extension 102
192.168.1.15 is the Asterisk server.
(I replaced my domain name per “my.domain” in the logs)

This is the end of the debug logs:

<— SIP read from UDP:192.168.1.72:50541 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK65f69116;rport=5163
To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975
From: “Cambridge” sip:121@my.domain;tag=as693d29d7
Call-ID: 4995bf7656e2875e347770af0109c756@my.domain
CSeq: 102 INVITE
Contact: sip:102@192.168.1.72:50541;transport=udp
Server: Sipdroid/1.5.7 beta/HTC Hero
Content-Length: 189
Content-Type: application/sdp

v=0
o=102@my.domain> 0 0 INIP4 192.168.1.72
s=Session SIP/SDP
c=IN IP4 192.168.1.72
t=0 0
m=audio 21000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

<------------->
— (10 headers 8 lines) —
list_route: hop: sip:102@192.168.1.72:50541;transport=udp
set_destination: Parsing sip:102@192.168.1.72:50541;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.72, port 50541
Transmitting (no NAT) to 192.168.1.72:50541:
ACK sip:102@192.168.1.72:50541;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK6f399895;rport
Max-Forwards: 70
From: “Cambridge” sip:121@my.domain;tag=as693d29d7
To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975
Contact: sip:121@192.168.1.15:5163
Call-ID: 4995bf7656e2875e347770af0109c756@my.domain
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

Audio is at 192.168.1.15 port 10086
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.18:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603
To: sip:102@my.domain;tag=as65ce952f
Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.15:5163
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1949615126 1949615127 IN IP4 192.168.1.15
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.15
t=0 0
m=audio 10086 RTP/AVP 0 99
a=rtpmap:0 PCMU/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

Thanks for your help,
Fabien

5

Hi again,
Also I did a tcpdump, and I can see the following:

Nortel -> asterisk INVITE
Phone -> Asterisk Status 180 Ringing
Asterisk -> Nortel Status 180 Ringing
Asterisk -> Nortel status 183 Session Progress
Then RTP traffic is sent from Nortel to the phone via Asterisk (in this way only):
Then RTP traffic is also sent from the phone to the Nortel via Asterisk
And later, the phone sends a SIP OK 200
but the Nortel still rings…

Thanks,
Fabien

6

The problem is with the Nortel.

7

Hi David,
Thing is that the problem does not occur with all the phones. It only happen when I use a SIP trunk, or to a few phones.
Is there any particular setting in Asterisk that could help solving this issue ? (like wait for 200 SIP before sending RTP traffic)
Thanks,
Fabien

8

Sorry. I am being confused about the directions here. The problem, I think, is that you have said rings, when you actually mean receives ringback tone. Also, your tcpdump summary is missing the INVITE from Asterisk to the Phone, which made me think that the 180 and 183 where associated with the original INVITE.

Asterisk should not be initiating early media unless the dialplan tells it to, or the called party requests it. In the latest versions, it shouldn’t initiate it unless the dialplan explicitly requests it (by using the Progress() application).

Asterisk should be sending 200 OK when it receives 200 OK from the phone. It appears to be doing that:

<— Reliably Transmitting (no NAT) to 192.168.1.18:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18

At that point it should cease sending any internally generated early media. You need the complete debug log to see whether Asterisk was trying to source ringback tone, or whether it is just passing it through.

There are options to disable the use of early media.

9

Hi,

Here is a trace :

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bd9-e961bad;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603
To: sip:102@my.domain;tag=as701ad038
Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603
To: sip:102@my.domain
Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK65f69116;rport=5163
To: sip:102@192.168.1.72:50541;transport=udp
From: “home” sip:121@my.domain;tag=as693d29d7
Call-ID: 4995bf7656e2875e347770af0109c756@my.domain

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK65f69116;rport=5163
To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975
From: “home” sip:121@my.domain;tag=as693d29d7
Call-ID: 4995bf7656e2875e347770af0109c756@my.domain

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603
To: sip:102@my.domain;tag=as65ce952f
Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603
To: sip:102@my.domain;tag=as65ce952f
Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK65f69116;rport=5163
To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975
From: “home” sip:121@my.domain;tag=as693d29d7
Call-ID: 4995bf7656e2875e347770af0109c756@my.domain

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603
To: sip:102@my.domain;tag=as65ce952f
Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-615-17c433-a626eb1;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603
To: sip:102@my.domain;tag=as65ce952f
Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK0ba736bc;rport=5163
To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975
From: “home” sip:121@my.domain;tag=as693d29d7
Call-ID: 4995bf7656e2875e347770af0109c756@my.domain

Thanks,
regards,
Fabien

10

That’s a very incomplete trace. There is only one side. The SIP headers are incomplete and there is no SDP payload. There is no Asterisk debug output showing what it is doing with the messages. It’s not worth my time looking at it in any detail.

11

Hi David,
I tried to cut the last trace, sorry about that.
Here is a complete trace :

Nortel phone at 192.168.1.18 gives a ring to phone at 192.168.1.72. Asterisk is at 192.168.1.15.
When the phone picks up, the Nortel phone keeps on ringing, then the phone hangs up.
Weird thing is I can see the first SIP 200 OK coming from the Nortel.

<— SIP read from UDP:192.168.1.18:5060 —>
INVITE sip:102@my.domain SIP/2.0
From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
To: sip:102@my.domain
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba50-3292b1a4
Max-Forwards: 70
Supported: replaces
User-Agent: Nortel IP Phone 1535 (0.2.50.0905)
Contact: sip:121@192.168.1.18:5060
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Content-Type: application/sdp
Content-Length: 400

v=0
o=LGEIPP 5762 5762 IN IP4 192.168.1.18
s=SIP Call
c=IN IP4 192.168.1.18
b=CT:160
t=0 0
m=audio 23000 RTP/AVP 0 8 18 4 95 99 111
c=IN IP4 192.168.1.18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:95 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=ptime:20
a=sendrecv

<------------->
— (13 headers 19 lines) —
Sending to 192.168.1.18 : 5060 (NAT)
Using INVITE request as basis request - 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
Found peer ‘121’ for ‘121’ from 192.168.1.18:5060

<— Reliably Transmitting (no NAT) to 192.168.1.18:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba50-3292b1a4;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
To: sip:102@my.domain;tag=as2885dfe9
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="65123c99"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.18:5060 —>
ACK sip:102@my.domain SIP/2.0
From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
To: sip:102@my.domain;tag=as2885dfe9
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba50-3292b1a4
Max-Forwards: 70
User-Agent: Nortel IP Phone 1535 (0.2.50.0905)
Contact: sip:121@192.168.1.18:5060
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:192.168.1.18:5060 —>
INVITE sip:102@my.domain SIP/2.0
From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
To: sip:102@my.domain
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a
Max-Forwards: 70
Supported: replaces
User-Agent: Nortel IP Phone 1535 (0.2.50.0905)
Contact: sip:121@192.168.1.18:5060
Authorization: Digest username=“121”,realm=“asterisk”,nonce=“65123c99”,uri=“sip:102@my.domain”,response=“d43d93964059735a8a1afeab3c96bbb3”,algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Content-Type: application/sdp
Content-Length: 400

v=0
o=LGEIPP 5762 5762 IN IP4 192.168.1.18
s=SIP Call
c=IN IP4 192.168.1.18
b=CT:160
t=0 0
m=audio 23000 RTP/AVP 0 8 18 4 95 99 111
c=IN IP4 192.168.1.18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:95 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=ptime:20
a=sendrecv

<------------->
— (14 headers 19 lines) —
Sending to 192.168.1.18 : 5060 (no NAT)
Using INVITE request as basis request - 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
Found peer ‘121’ for ‘121’ from 192.168.1.18:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 95
Found RTP audio format 99
Found RTP audio format 111
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format AMR for ID 95
Found audio description format telephone-event for ID 99
Found audio description format X-nt-inforeq for ID 111
Capabilities: us - 0x380004 (ulaw|h263|h263p|h264), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.18:23000
Peer doesn’t provide video
Looking for 102 in DLPN_DialPlan1 (domain my.domain)
list_route: hop: sip:121@192.168.1.18:5060

<— Transmitting (no NAT) to 192.168.1.18:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
To: sip:102@my.domain
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.15:5163
Content-Length: 0

<------------>
Audio is at 192.168.1.15 port 10078
Video is at 192.168.1.15 port 10070
Adding codec 0x4 (ulaw) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding video codec 0x100000 (h263p) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.72:34702:
INVITE sip:102@192.168.1.72:34702;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK3fd058b0;rport
Max-Forwards: 70
From: “home” sip:121@my.domain;tag=as53a84d81
To: sip:102@192.168.1.72:34702;transport=udp
Contact: sip:121@192.168.1.15:5163
Call-ID: 64dc54dc7366e87d6d06cef8780b5cfa@my.domain
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 22 Sep 2010 21:50:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 393

v=0
o=root 234067187 234067187 IN IP4 192.168.1.15
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.15
b=CT:384
t=0 0
m=audio 10078 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10070 RTP/AVP 34 98 99
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv

<— SIP read from UDP:192.168.1.72:34702 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK3fd058b0;rport=5163
To: sip:102@192.168.1.72:34702;transport=udp
From: “home” sip:121@my.domain;tag=as53a84d81
Call-ID: 64dc54dc7366e87d6d06cef8780b5cfa@my.domain
CSeq: 102 INVITE
Server: Sipdroid/1.5.7 beta/HTC Hero
Content-Length: 0

<— SIP read from UDP:192.168.1.72:34702 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK3fd058b0;rport=5163
To: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f
From: “home” sip:121@my.domain;tag=as53a84d81
Call-ID: 64dc54dc7366e87d6d06cef8780b5cfa@my.domain
CSeq: 102 INVITE
Server: Sipdroid/1.5.7 beta/HTC Hero
Content-Length: 189
Content-Type: application/sdp

v=0
o=102@my.domain 0 0 IN IP4 192.168.1.72
s=Session SIP/SDP
c=IN IP4 192.168.1.72
t=0 0
m=audio 21000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

<------------->
— (9 headers 8 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x380004 (ulaw|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.72:21000
Peer doesn’t provide video

<— Transmitting (no NAT) to 192.168.1.18:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
To: sip:102@my.domain;tag=as4162285c
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.15:5163
Content-Length: 0

<------------>
Audio is at 192.168.1.15 port 10076
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 192.168.1.18:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
To: sip:102@my.domain;tag=as4162285c
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.15:5163
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 954720937 954720937 IN IP4 192.168.1.15
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.15
t=0 0
m=audio 10076 RTP/AVP 0 99
a=rtpmap:0 PCMU/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:192.168.1.72:34702 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK3fd058b0;rport=5163
To: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f
From: “home” sip:121@my.domain;tag=as53a84d81
Call-ID: 64dc54dc7366e87d6d06cef8780b5cfa@my.domain
CSeq: 102 INVITE
Contact: sip:102@192.168.1.72:34702;transport=udp
Server: Sipdroid/1.5.7 beta/HTC Hero
Content-Length: 189
Content-Type: application/sdp

v=0
o=102@my.domain 0 0 IN IP4 192.168.1.72
s=Session SIP/SDP
c=IN IP4 192.168.1.72
t=0 0
m=audio 21000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

<------------->
— (10 headers 8 lines) —
list_route: hop: sip:102@192.168.1.72:34702;transport=udp
set_destination: Parsing sip:102@192.168.1.72:34702;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.72, port 34702
Transmitting (no NAT) to 192.168.1.72:34702:
ACK sip:102@192.168.1.72:34702;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK7b855d14;rport
Max-Forwards: 70
From: “home” sip:121@my.domain;tag=as53a84d81
To: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f
Contact: sip:121@192.168.1.15:5163
Call-ID: 64dc54dc7366e87d6d06cef8780b5cfa@my.domain
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

Audio is at 192.168.1.15 port 10076
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.18:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a;received=192.168.1.18
From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
To: sip:102@my.domain;tag=as4162285c
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.15:5163
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 954720937 954720938 IN IP4 192.168.1.15
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.15
t=0 0
m=audio 10076 RTP/AVP 0 99
a=rtpmap:0 PCMU/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>

<— SIP read from UDP:192.168.1.18:5060 —>
ACK sip:102@192.168.1.15:5163 SIP/2.0
From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
To: sip:102@my.domain;tag=as4162285c
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d26-1e8df86-59f5a8b6
Max-Forwards: 70
User-Agent: Nortel IP Phone 1535 (0.2.50.0905)
Contact: sip:121@192.168.1.18:5060
Authorization: Digest username=“121”,realm=“asterisk”,nonce=“65123c99”,uri=“sip:102@my.domain”,response=“d43d93964059735a8a1afeab3c96bbb3”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Reliably Transmitting (no NAT) to 192.168.1.72:34702:
OPTIONS sip:102@192.168.1.72:34702;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK32634603;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@my.domain;tag=as11fdd8cb
To: sip:102@192.168.1.72:34702;transport=udp
Contact: sip:asterisk@192.168.1.15:5163
Call-ID: 576d57b62e7f8dfd37ca1790325bfd77@my.domain
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 22 Sep 2010 21:50:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.72:34702 —>
BYE sip:121@192.168.1.15:5163 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.72:34702;rport;branch=z9hG4bK39294
Max-Forwards: 70
To: “home” sip:121@my.domain;tag=as53a84d81
From: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f
Call-ID: 64dc54dc7366e87d6d06cef8780b5cfa@my.domain
CSeq: 1 BYE
User-Agent: Sipdroid/1.5.7 beta/HTC Hero
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.72 : 34702 (no NAT)

<— Transmitting (no NAT) to 192.168.1.72:34702 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.72:34702;branch=z9hG4bK39294;received=192.168.1.72;rport=34702
From: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f
To: “home” sip:121@my.domain;tag=as53a84d81
Call-ID: 64dc54dc7366e87d6d06cef8780b5cfa@my.domain
CSeq: 1 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d’ in 32000 ms (Method: ACK)
set_destination: Parsing sip:121@192.168.1.18:5060 for address/port to send to
set_destination: set destination to 192.168.1.18, port 5060
Reliably Transmitting (no NAT) to 192.168.1.18:5060:
BYE sip:121@192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK1aa94777;rport
Max-Forwards: 70
From: sip:102@my.domain;tag=as4162285c
To: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 102 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Really destroying SIP dialog '64dc54dc7366e87d6d06cef8780b5cfa@my.domain’ Method: BYE

<— SIP read from UDP:192.168.1.18:5060 —>
SIP/2.0 200 OK
From: sip:102@my.domain;tag=as4162285c
To: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d
Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d
CSeq: 102 BYE
Via: SIP/2.0/UDP 192.168.1.15:5163;rport=5163;branch=z9hG4bK1aa94777
Supported: replaces
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
User-Agent: Nortel IP Phone 1535 (0.2.50.0905)
Content-Length: 0

<------------->
— (10 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d’ Method: ACK
Retransmitting #1 (no NAT) to 192.168.1.72:34702:
OPTIONS sip:102@192.168.1.72:34702;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK32634603;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@my.domain;tag=as11fdd8cb
To: sip:102@192.168.1.72:34702;transport=udp
Contact: sip:asterisk@192.168.1.15:5163
Call-ID: 576d57b62e7f8dfd37ca1790325bfd77@my.domain
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 22 Sep 2010 21:50:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.72:34702 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK32634603;rport=5163
To: sip:102@192.168.1.72:34702;transport=udp
From: “asterisk” sip:asterisk@my.domain;tag=as11fdd8cb
Call-ID: 576d57b62e7f8dfd37ca1790325bfd77@my.domain
CSeq: 102 OPTIONS
Contact: sip:102@192.168.1.72:34702;transport=udp
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog '576d57b62e7f8dfd37ca1790325bfd77@my.domain’ Method: OPTIONS

<— SIP read from UDP:192.168.1.72:34702 —>

<------------->

Is it a issue with early media ? How can I remove it if it is the case ? Adding “progress …” in the Dial function ?

Thanks for your help,
Regards,

Fabien

12

Hi David,

I think I found a solution (thanks to your comments).
I have added “r” in the Dial function in extensions.conf (exten = 1-dial,1,Dial(${ARG1},r)
and now the Nortel stops ringing when the other sides picks-up, and the communication works fine.

From the documentation of the Dial function:
r — indicate call ringing to the calling extension

This option is used when your phone set doesn’t recognize the traditional call progress indications. The best example of this is when you don’t hear the “ringing” on the phone when you call another extension. With the ‘r’ option turned on, ring tone is passed as actual audio on the phone line, allowing you to hear the call progress.

Is this something that can be a problem as the parameter that I modified apply for every phones ?
The solution may be to create a special DialPlan but is there a way to modify the variable “DIALOPTIONS” depending of the extension that make the call ?

Thanks,
Fabien

13

I also have this problem with PIAF, running asterisk v 1.6x. Any idea of how to fix this using the FreePBX interface?

14

I have this problem too, running PIAF, using multiple Nortel 1535 phones, calling a variety of numbers. I have tried a number of things, no of which seem to have any effect on the problem:

  • Upgraded phone firmware from 0.2.50 to 0.2.76
  • Added “r” to dialout options, and verified via log that it is using this:

[2011-07-01 12:11:32] VERBOSE[5743] pbx.c: – Executing [s@macro-dialout-trunk:19] Dial(“SIP/111-00000007”, “SIP/VP-SIPSJCA/+14155551212,300,r”) in new stack

  • Running Asterisk 1.8.2.3
  • FreePBX 2.8.1
  • Using VoicePulse

Obviously this phone must work better for most people, or I would hear more complaints. Is it possible that this situation is specifically worse with VoicePulse?