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Summary:ASTERISK-25457: Chan_PJSIP No MoH / Hold
Reporter:Ross Beer (rossbeer)Labels:
Date Opened:2015-10-09 05:21:38Date Closed:2016-02-02 03:19:21.000-0600
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_pjsip
Versions:13.6.0 Frequency of
Occurrence
Related
Issues:
Environment:Snom 7XX, SNOM 3XX, CISCO SPAXXXXAttachments:
Description: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}
Comments:By: Asterisk Team (asteriskteam) 2015-10-09 05:21:39.885-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Rusty Newton (rnewton) 2015-11-06 12:47:21.522-0600

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@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@10.24.22.51;line=1tzct067>;tag=qkbzytune8
To: "BOB" <sip:BOB@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@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@10.24.22.51;line=1tzct067>;tag=qkbzytune8
To: "BOB" <sip:BOB@10.24.18.124>;tag=2d1e8304-57d7-4c99-8a5c-831bcc5fdd42
CSeq: 1 INVITE
Session-Expires: 1800;refresher=uas
Contact: <sip:asterisk@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@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@10.24.22.51;line=1tzct067>;tag=qkbzytune8
To: "BOB" <sip:BOB@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@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?
By: Ross Beer (rossbeer) 2015-11-16 08:17:35.167-0600

I have tried using the Asterisk 13 (GIT-13-506aea2 , beyond 13.6.0) version and I can confirm that the issue is not present in this version only in the general release 13.6.0.

Will the changes from the GIT repository be merged into the next release?
By: Rusty Newton (rnewton) 2015-11-18 16:01:41.460-0600

bq. Will the changes from the GIT repository be merged into the next release?

Yes they will. I'll go ahead and close this out then. Thanks for reporting back.

Also I'll mention the other issue that I found in case it is of use to you. When calling the Snom phone it fails to answer if the *display name* section of the From URI in the INVITE contains a format like "name <number>". Specifically the greater than and less than signs. When receiving a call as described the Snom does not respond upon lifting the handset or pressing the answer button. Very strange! I reported this bug to Snom.
By: Ross Beer (rossbeer) 2016-01-26 07:55:06.807-0600

What is the best way to identify the changes between Asterisk 13 (GIT-13-506aea2 , beyond 13.6.0) and Asterisk release 13.7.0?
By: Ross Beer (rossbeer) 2016-02-02 03:19:06.281-0600

This issue looks to be related to the default MoH class not being played when a class is defined but not found.

The trunk version plays the default MoH in this situation, but the release doesn't.