[asterisk-bugs] [JIRA] (ASTERISK-25457) Chan_PJSIP No MoH / Hold
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Fri Nov 6 12:49:33 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-25457?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228198#comment-228198 ]
Rusty Newton edited comment on ASTERISK-25457 at 11/6/15 12:47 PM:
-------------------------------------------------------------------
I can't reproduce.
I setup a Snom 715 (snom715-SIP 8.7.5.8) with Asterisk 13 (GIT-13-506aea2 , beyond 13.6.0)
Hold works from the Snom. I tried calling a few different directions and it always seems to work.
my pjsip.conf is:
{noformat}
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
[ALICE]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=ALICE
aors=ALICE
direct_media=no
[ALICE]
type=auth
auth_type=userpass
password=ALICE
username=ALICE
[ALICE]
type=aor
max_contacts=1
{noformat}
All other extensions were setup identically. All extensions including the Snom were able to initiate hold and able to hear MOH when on the held side.
An example of the Snom hold initiation:
{noformat}
<--- Received SIP request (1431 bytes) from UDP:10.24.22.51:46581 --->
INVITE sip:asterisk at 10.24.18.124:5060 SIP/2.0
Via: SIP/2.0/UDP 10.24.22.51:46581;branch=z9hG4bK-xtkli27o7hlk;rport
From: <sip:CAROL at 10.24.22.51;line=1tzct067>;tag=qkbzytune8
To: "BOB" <sip:BOB at 10.24.18.124>;tag=2d1e8304-57d7-4c99-8a5c-831bcc5fdd42
Call-ID: ea6ce005-79e1-4cd3-97c3-ec262fca0753
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom715/8.7.5.8
Contact: <sip:CAROL at 10.24.22.51:46581;line=1tzct067>;reg-id=1
X-Serialnumber: 000413750969
P-Key-Flags: resolution="31x13", keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 598
v=0
o=root 1533464206 1533464208 IN IP4 10.24.22.51
s=call
c=IN IP4 10.24.22.51
t=0 0
m=audio 49204 RTP/AVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:112 AAL2-G726-16/8000
a=rtpmap:113 AAL2-G726-24/8000
a=rtpmap:114 AAL2-G726-32/8000
a=rtpmap:115 AAL2-G726-40/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
<--- Transmitting SIP response (867 bytes) to UDP:10.24.22.51:46581 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.24.22.51:46581;rport=46581;received=10.24.22.51;branch=z9hG4bK-xtkli27o7hlk
Call-ID: ea6ce005-79e1-4cd3-97c3-ec262fca0753
From: <sip:CAROL at 10.24.22.51;line=1tzct067>;tag=qkbzytune8
To: "BOB" <sip:BOB at 10.24.18.124>;tag=2d1e8304-57d7-4c99-8a5c-831bcc5fdd42
CSeq: 1 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk at 10.24.18.124:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Server: MyUserAgentValue
Content-Type: application/sdp
Content-Length: 235
v=0
o=- 497631372 497631373 IN IP4 10.24.18.124
s=Asterisk
c=IN IP4 10.24.18.124
t=0 0
m=audio 19300 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=recvonly
-- Started music on hold, class 'default', on channel 'PJSIP/BOB-00000000'
<--- Received SIP request (440 bytes) from UDP:10.24.22.51:46581 --->
ACK sip:asterisk at 10.24.18.124:5060 SIP/2.0
Via: SIP/2.0/UDP 10.24.22.51:46581;branch=z9hG4bK-jjx3evnsem4a;rport
From: <sip:CAROL at 10.24.22.51;line=1tzct067>;tag=qkbzytune8
To: "BOB" <sip:BOB at 10.24.18.124>;tag=2d1e8304-57d7-4c99-8a5c-831bcc5fdd42
Call-ID: ea6ce005-79e1-4cd3-97c3-ec262fca0753
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom715/8.7.5.8
Contact: <sip:CAROL at 10.24.22.51:46581;line=1tzct067>;reg-id=1
Content-Length: 0
{noformat}
So, I can't reproduce that issue. However I did find a different issue where there is one-way audio when the Snom calls Digium phones which is strange. I'll file a separate issue for that.
Do you have any further information on how to reproduce your hold/MOH issue?
was (Author: rnewton):
I can't reproduce.
I setup a Snom 715 (snom715-SIP 8.7.5.8) with Asterisk 13 (GIT-13-506aea2 , beyond 13.6.0)
Hold works from the Snom. I tried calling a few different directions and it always seems to work.
my pjsip.conf is:
{noformat}
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
[ALICE]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=ALICE
aors=ALICE
direct_media=no
[ALICE]
type=auth
auth_type=userpass
password=ALICE
username=ALICE
[ALICE]
type=aor
max_contacts=1
{noformat}
All other extensions were setup identically. All extensions including the Snom were able to initiate hold and able to hear MOH when on the held side.
An example of the Snom hold initiation:
{noformat}
<--- Received SIP request (1431 bytes) from UDP:10.24.22.51:46581 --->
INVITE sip:asterisk at 10.24.18.124:5060 SIP/2.0
Via: SIP/2.0/UDP 10.24.22.51:46581;branch=z9hG4bK-xtkli27o7hlk;rport
From: <sip:CAROL at 10.24.22.51;line=1tzct067>;tag=qkbzytune8
To: "BOB" <sip:BOB at 10.24.18.124>;tag=2d1e8304-57d7-4c99-8a5c-831bcc5fdd42
Call-ID: ea6ce005-79e1-4cd3-97c3-ec262fca0753
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom715/8.7.5.8
Contact: <sip:CAROL at 10.24.22.51:46581;line=1tzct067>;reg-id=1
X-Serialnumber: 000413750969
P-Key-Flags: resolution="31x13", keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Require: timer
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 598
v=0
o=root 1533464206 1533464208 IN IP4 10.24.22.51
s=call
c=IN IP4 10.24.22.51
t=0 0
m=audio 49204 RTP/AVP 9 0 8 3 97 98 99 100 112 113 114 115 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:112 AAL2-G726-16/8000
a=rtpmap:113 AAL2-G726-24/8000
a=rtpmap:114 AAL2-G726-32/8000
a=rtpmap:115 AAL2-G726-40/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
<--- Transmitting SIP response (867 bytes) to UDP:10.24.22.51:46581 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.24.22.51:46581;rport=46581;received=10.24.22.51;branch=z9hG4bK-xtkli27o7hlk
Call-ID: ea6ce005-79e1-4cd3-97c3-ec262fca0753
From: <sip:CAROL at 10.24.22.51;line=1tzct067>;tag=qkbzytune8
To: "BOB" <sip:BOB at 10.24.18.124>;tag=2d1e8304-57d7-4c99-8a5c-831bcc5fdd42
CSeq: 1 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk at 10.24.18.124:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Supported: 100rel, timer, replaces, norefersub
Server: MyUserAgentValue
Content-Type: application/sdp
Content-Length: 235
v=0
o=- 497631372 497631373 IN IP4 10.24.18.124
s=Asterisk
c=IN IP4 10.24.18.124
t=0 0
m=audio 19300 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=recvonly
-- Started music on hold, class 'default', on channel 'PJSIP/BOB-00000000'
<--- Received SIP request (440 bytes) from UDP:10.24.22.51:46581 --->
ACK sip:asterisk at 10.24.18.124:5060 SIP/2.0
Via: SIP/2.0/UDP 10.24.22.51:46581;branch=z9hG4bK-jjx3evnsem4a;rport
From: <sip:CAROL at 10.24.22.51;line=1tzct067>;tag=qkbzytune8
To: "BOB" <sip:BOB at 10.24.18.124>;tag=2d1e8304-57d7-4c99-8a5c-831bcc5fdd42
Call-ID: ea6ce005-79e1-4cd3-97c3-ec262fca0753
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom715/8.7.5.8
Contact: <sip:CAROL at 10.24.22.51:46581;line=1tzct067>;reg-id=1
Content-Length: 0
{noformat}
So, I can't reproduce that issue. However I did find a different issue where there is one-way audio when calling Digium phones which is strange. I'll file a separate issue for that.
Do you have any further information on how to reproduce your hold/MOH issue?
> Chan_PJSIP No MoH / Hold
> ------------------------
>
> Key: ASTERISK-25457
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25457
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_pjsip
> Affects Versions: 13.6.0
> Environment: Snom 7XX, SNOM 3XX, CISCO SPAXXXX
> Reporter: Ross Beer
> Assignee: Ross Beer
> Severity: Minor
>
> When using PJSIP with Snom and Cisco SPA phones MoH is not initiated. MoH works fine with queues etc.
> I believe the issue is that the phones are setting the 'sendonly' header on the invite instead of an IP address of 0.0.0.0. The 'sendonly' method does however work with chan_sip for all devices.
> {noformat}
> INVITE sip:<IPADDRESS>:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.104:8400;branch=z9hG4bK-5w3f4vtw2br5;rport
> From: "<NUMBER>" <sip:<USERNAME>@<IPADDRESS>>;tag=4fwufutws8
> To: <sip:<NUMBER>@<IPADDRESS>;user=phone>;tag=3e9c3595-e535-47fc-a5b2-d1e241b2c7b3
> Call-ID: 3134343433383534313932353436-c7z7ca4mvv41
> CSeq: 3 INVITE
> Max-Forwards: 70
> User-Agent: snom760/8.7.5.28
> Contact: <sip:<USERNAME>@192.168.100.104:8400>;reg-id=1
> X-Serialnumber: 000413710A91
> P-Key-Flags: resolution="31x13", keys="4"
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, replaces, from-change
> Session-Expires: 3600;refresher=uac
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 271
> v=0
> o=root 358968259 358968260 IN IP4 192.168.100.104
> s=call
> c=IN IP4 192.168.100.104
> t=0 0
> m=audio 65284 RTP/AVP 8 0 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendonly
> {noformat}
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