Call Pickup On Channel Sip With SNOM Phones Issue

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Hi,

while trying to get my new Asterisk 15.5.0 PBX replacing a 11 years old Asterisk 1.2.31 ISDN BPX, I’m stuck to get call pickup going as usual. The old one uses specific patches, IIRC…

If I interpret various sources of related information correctly, current Asterisk versions should support this feature out of the box.

According to http://wiki.snom.com/Category:HowTo:Call_Pickup, there are several ways to get this feature going. I’m enjoying method (1) since ages, but I couldn’t get asterisk to send the full NOTIFY xml dialog-info including call-id, remote and local values, although setting context = phones allowsubscribe = yes subscribecontext = phones notifyringing = yes notifycid = ignore-context

as well as callgroup = 1
pickupgroup = 1

for every local phone (all snom, mostly 360 phones) in sip.conf.

[phones]
exten => 60,hint,SIP/60
exten => 61,hint,SIP/61
exten => 62,hint,SIP/62

exten => _60,1,Dial(SIP/60)
exten => _61,1,Dial(SIP/61)
exten => _62,1,Dial(SIP/62)

but the notify looks like this:

—
== Extension Changed 62[phones] new state Ringing for Notify User 62
Reliably Transmitting (NAT) to 172.16.23.60:2112:
NOTIFY sip:60@172.16.23.60:2112 SIP/2.0
Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK7d7c5ac4;rport Max-Forwards: 70
From: ;tag=as72f9c98b To: ;tag=ufln5vo7x5
Contact:
Call-ID: 3c94f02212d4-8we7ggt625fi CSeq: 106 NOTIFY
User-Agent: Asterisk PBX 15.5.0
Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 217




early

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

<--- SIP read from UDP:172.16.23.60:2112 --->
SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK7d7c5ac4;rport=5060
From: ;tag=as72f9c98b To: ;tag=ufln5vo7x5
Call-ID: 3c94f02212d4-8we7ggt625fi CSeq: 106 NOTIFY
Content-Length: 0

SIP/60 is notified correctly, but misses the notifycid information. I’ve tried both, notifycid = yes and notifycid = ignore-context of course.

*CLI> core show hints
62@phones : SIP/62 State:Idle Presence:not_set Watchers 3
61@phones : SIP/61 State:Idle Presence:not_set Watchers 2
60@phones : SIP/60 State:Idle Presence:not_set Watchers 3

*CLI> sip show subscriptions Peer User Call ID Extension Last state Type Mailbox Expiry
172.16.23.60 60 3c94f0220769-0p 61@phones Idle dialog-info+xml 003600
172.16.23.60 60 3c94f0220182-27 60@phones Idle dialog-info+xml 003600
172.16.23.62 62 313533353338323 60@phones Idle dialog-info+xml 003600
172.16.23.62 62 313533353338323 62@phones Idle dialog-info+xml 003600
172.16.23.60 60 3c94f0220d01-p2 62@phones Idle dialog-info+xml 003600
172.16.23.60 60 3c94f023462b-j2 60@phones Idle dialog-info+xml 003600
172.16.23.60 60 3c94f02212d4-8w 62@phones Idle dialog-info+xml 003600
172.16.23.62 62 313533353338323 61@phones Idle dialog-info+xml 003600

What am I missing here, any suggestions?

Cheers, Pete

—

One thought on - Call Pickup On Channel Sip With SNOM Phones Issue

  • Okay, scratch it, “notifycid = yes” must reside in the general section!

    Now, it behaves as expected until:

    [Aug 27 22:20:37] NOTICE[6200][C-00000003]: app_directed_pickup.c:365 pickup_exec: No target channel found for 62@phones

    Details:

    extensions.conf:

    [phones]
    exten => 60,hint,SIP/60
    exten => 61,hint,SIP/61
    exten => 62,hint,SIP/62

    exten => _60,1,Dial(SIP/60)
    exten => _61,1,Dial(SIP/61)
    exten => _62,1,Dial(SIP/62)

    A call from external to 62 is notified to 60 three times:

    First a little silly (local and remote are identical):

    == Extension Changed 62[phones] new state Ringing for Notify User 60
    Reliably Transmitting (NAT) to 172.16.23.60:2112:
    NOTIFY sip:60@172.16.23.60:2112 SIP/2.0
    Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK2ec2ef2e;rport Max-Forwards: 70
    From: ;tag=as40973611
    To: ;tag=ebsb74m178
    Contact:
    Call-ID: 3c95372c15b1-uz42rw4w6sy9
    CSeq: 104 NOTIFY
    User-Agent: Asterisk PBX 15.5.0
    Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 524





    sip:62@172.16.4.100



    sip:62@172.16.4.100


    early

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

    Then with an entity of sip:62@172.16.23.8, which is my old asterisk, but with correct local/remote values:

    — (7 headers 0 lines) —
    Reliably Transmitting (NAT) to 172.16.23.60:2112:
    NOTIFY sip:60@172.16.23.60:2112 SIP/2.0
    Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK60e3a3a9;rport Max-Forwards: 70
    From: ;tag=as2c6b3fce To: ;tag=mafy78cezc Contact:
    Call-ID: 3c95372c1b80-xmqzyr2cq6z2
    CSeq: 105 NOTIFY
    User-Agent: Asterisk PBX 15.5.0
    Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 541





    sip:01721234567@172.16.4.100



    sip:62@172.16.4.100


    early

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

    And finally correctly:

    — (7 headers 0 lines) —
    Reliably Transmitting (NAT) to 172.16.23.60:2112:
    NOTIFY sip:60@172.16.23.60:2112 SIP/2.0
    Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK6d9e9f60;rport Max-Forwards: 70
    From: ;tag=as40973611
    To: ;tag=ebsb74m178
    Contact:
    Call-ID: 3c95372c15b1-uz42rw4w6sy9
    CSeq: 105 NOTIFY
    User-Agent: Asterisk PBX 15.5.0
    Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 542





    sip:01721234567@172.16.4.100



    sip:62@172.16.4.100


    early

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

    NOTIFY Ack:

    — (11 headers 0 lines) —
    sip_route_dump: route/path hop:
    — SIP/62-00000003 is ringing

    <--- SIP read from UDP:172.16.23.60:2112 --->
    SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK6d9e9f60;rport=5060
    From: ;tag=as40973611
    To: ;tag=ebsb74m178
    Call-ID: 3c95372c15b1-uz42rw4w6sy9
    CSeq: 105 NOTIFY
    Content-Length: 0

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

    60 want to take over the call:

    <--- SIP read from UDP:172.16.23.60:2112 --->
    INVITE sip:01721234567@172.16.4.100 SIP/2.0
    Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-pb8ywn90vhet;rport From: “HFO” ;tag=omy5lrfdik To:
    Call-ID: 3c953745720b-p5q7kgcj604q CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: ;reg-id=1
    Replaces: pickup-3c95372c15b1-uz42rw4w6sy9;to-tag=as40973611;from-tag=ebsb74m178
    P-Key-Flags: resolution=”31×13″, keys=”4″
    User-Agent: snom360/7.3.30
    Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
    Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90
    Content-Type: application/sdp Content-Length: 452

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

    Unauthorized:

    — (19 headers 18 lines) —
    Sending to 172.16.23.60:2112 (NAT)
    [Aug 27 22:20:37] NOTICE[6200][C-00000003]: chan_sip.c:26269 handle_request_invite: Trying to pick up 62@phones Sending to 172.16.23.60:2112 (NAT)
    Using INVITE request as basis request – 3c953745720b-p5q7kgcj604q Found peer ’60’ for ’60’ from 172.16.23.60:2112

    <--- Reliably Transmitting (NAT) to 172.16.23.60:2112 --->
    SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-pb8ywn90vhet;received=172.16.23.60;rport=2112
    From: “HFO” ;tag=omy5lrfdik To: ;tag=as36a783db Call-ID: 3c953745720b-p5q7kgcj604q CSeq: 1 INVITE
    Server: Asterisk PBX 15.5.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=”asterisk”, nonce=”5835824d”
    Content-Length: 0

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

    Ahh, okay

    <--- SIP read from UDP:172.16.23.60:2112 --->
    ACK sip:01721234567@172.16.4.100 SIP/2.0
    Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-pb8ywn90vhet;rport From: “HFO” ;tag=omy5lrfdik To: ;tag=as36a783db Call-ID: 3c953745720b-p5q7kgcj604q CSeq: 1 ACK
    Max-Forwards: 70
    Contact: ;reg-id=1
    Content-Length: 0

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

    You want auth, you get auth:

    <--- SIP read from UDP:172.16.23.60:2112 --->
    INVITE sip:01721234567@172.16.4.100 SIP/2.0
    Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-zbjhw9im94ce;rport From: “HFO” ;tag=omy5lrfdik To:
    Call-ID: 3c953745720b-p5q7kgcj604q CSeq: 2 INVITE
    Max-Forwards: 70
    Contact: ;reg-id=1
    Replaces: pickup-3c95372c15b1-uz42rw4w6sy9;to-tag=as40973611;from-tag=ebsb74m178
    P-Key-Flags: resolution=”31×13″, keys=”4″
    User-Agent: snom360/7.3.30
    Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
    Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90
    Authorization: Digest username=”60″,realm=”asterisk”,nonce=”5835824d”,uri=”sip:01721234567@172.16.4.100″,response=”d7b5ae311ffa0c9f41154f952600220e”,algorithm=MD5
    Content-Type: application/sdp Content-Length: 452

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

    Figure out RTP parameter:

    — (20 headers 18 lines) —
    [Aug 27 22:20:37] NOTICE[6200][C-00000003]: chan_sip.c:26269 handle_request_invite: Trying to pick up 62@phones Sending to 172.16.23.60:2112 (NAT)
    Using INVITE request as basis request – 3c953745720b-p5q7kgcj604q Found peer ’60’ for ’60’ from 172.16.23.60:2112
    == Using SIP RTP CoS mark 5
    Found RTP audio format 9
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 3
    Found RTP audio format 18
    Found RTP audio format 4
    Found RTP audio format 99
    Found RTP audio format 101
    [Aug 27 22:20:37] NOTICE[6200][C-00000003]: chan_sip.c:10742 process_sdp: Processed audio crypto attribute without SAVP specified; accepting anyway Found audio description format g722 for ID 9
    Found audio description format pcmu for ID 0
    Found audio description format pcma for ID 8
    Found audio description format gsm for ID 3
    Found audio description format g729 for ID 18
    Found audio description format g723 for ID 4
    Found audio description format g726-32 for ID 99
    Found audio description format telephone-event for ID 101
    Capabilities: us – (alaw|ulaw|gsm), peer – audio=(ulaw|gsm|g723|alaw|g722|g729|g726)/video=(nothing)/text=(nothing), combined – (alaw|ulaw|gsm)
    Non-codec capabilities (dtmf): us – 0x0 (nothing), peer – 0x1 (telephone-event|), combined – 0x0 (nothing)
    > 0x7f0fc405ecf0 — Strict RTP learning after remote address set to: 172.16.23.60:54560
    Peer audio RTP is at port 172.16.23.60:54560
    Looking for 01721234567 in phones (domain 172.16.4.100)
    sip_route_dump: route/path hop:

    Why asterisk is looking up the external caller in phones?

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

    Let’s talk together:

    <--- Transmitting (NAT) to 172.16.23.60:2112 --->
    SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-zbjhw9im94ce;received=172.16.23.60;rport=2112
    From: “HFO” ;tag=omy5lrfdik To:
    Call-ID: 3c953745720b-p5q7kgcj604q CSeq: 2 INVITE
    Server: Asterisk PBX 15.5.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact:
    Content-Length: 0

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

    Here it goes south: No target channel found for 62@phones

    [Aug 27 22:20:37] NOTICE[6200][C-00000003]: app_directed_pickup.c:365 pickup_exec: No target channel found for 62@phones. Scheduling destruction of SIP dialog ‘3c953745720b-p5q7kgcj604q’ in 32000 ms (Method: INVITE)
    Reliably Transmitting (NAT) to 172.16.23.60:2112:
    NOTIFY sip:60@172.16.23.60:2112 SIP/2.0
    Via: SIP/2.0/UDP 172.16.4.100:5060;branch=z9hG4bK126d8508;rport Max-Forwards: 70
    From: ;tag=as06a1662e To: ;tag=9ce2167e53
    Contact:
    Call-ID: 3c95372c0a95-8efi7mvgrsud CSeq: 104 NOTIFY
    User-Agent: Asterisk PBX 15.5.0
    Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 200




    confirmed

    —

    Game over:

    <--- Reliably Transmitting (NAT) to 172.16.23.60:2112 --->
    SIP/2.0 603 Declined Via: SIP/2.0/UDP 172.16.23.60:2112;branch=z9hG4bK-zbjhw9im94ce;received=172.16.23.60;rport=2112
    From: “HFO” ;tag=omy5lrfdik To: ;tag=as19bd5d41
    Call-ID: 3c953745720b-p5q7kgcj604q CSeq: 2 INVITE
    Server: Asterisk PBX 15.5.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer Session-Expires: 1800;refresher=uas Content-Length: 0

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

    It should work as such, but doesn’t.

    Any ideas are much appreciated.

    Pete

    —