bearcapability_notavail

Asterisk Asterisk Support
1

Hi all

I am new to asterisk 1.8 recently I switched from asterisk 1.6.2.24 to asterisk 1.8.18.0 now I am communucating with asterisk 1.8 using asterisk-java-1.0.0.M3 and for conferenceing I am using app_konference 2.2.
when I try to create originate to conference channel hangsup and I get asterisk-java log as
02:26:20,802 INFO ChannelManager:590 - Removing channel SIP/0210-000000c4 due to hangup (BEARERCAPABILITY_NOTAVAIL)

and on asterisk sip debug i get following dump

<— SIP read from UDP:192.168.1.15:5060 —>
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;rport;branch=z9hG4bKPjXmsVI4ixZX3m9ME4f0ornq5wgQgWh64u
Max-Forwards: 70
From: sip:0260@192.168.0.3;tag=1MHx1fvpNNL5yQngeLpiA-EHIvy9sgEQ
To: sip:0260@192.168.0.3
Call-ID: Yz2wQMhmA6Nnc5MTCkNZIyCHJLI1EvEZ
CSeq: 9603 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: sip:0260@192.168.1.15:5060
Expires: 60
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.15:5060 (NAT)

<— Transmitting (NAT) to 192.168.1.15:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bKPjXmsVI4ixZX3m9ME4f0ornq5wgQgWh64u;received=192.168.1.15;rport=5060
From: sip:0260@192.168.0.3;tag=1MHx1fvpNNL5yQngeLpiA-EHIvy9sgEQ
To: sip:0260@192.168.0.3;tag=as1f0a38d2
Call-ID: Yz2wQMhmA6Nnc5MTCkNZIyCHJLI1EvEZ
CSeq: 9603 REGISTER
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="731a0158"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘Yz2wQMhmA6Nnc5MTCkNZIyCHJLI1EvEZ’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.15:5060 —>
REGISTER sip:192.168.0.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;rport;branch=z9hG4bKPjMTRAVX5m.FYhqd.bDGrAh17tGootLxK.
Max-Forwards: 70
From: sip:0260@192.168.0.3;tag=1MHx1fvpNNL5yQngeLpiA-EHIvy9sgEQ
To: sip:0260@192.168.0.3
Call-ID: Yz2wQMhmA6Nnc5MTCkNZIyCHJLI1EvEZ
CSeq: 9604 REGISTER
User-Agent: PJSUA v1.8/arm-none-linux-gnueabi
Contact: sip:0260@192.168.1.15:5060
Expires: 60
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username=“0260”, realm=“asterisk”, nonce=“731a0158”, uri=“sip:192.168.0.3”, response=“d9826fe08f8ca2e44f647a2b8eb96f22”, algorithm=MD5
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.15:5060 (NAT)

<— Transmitting (NAT) to 192.168.1.15:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bKPjMTRAVX5m.FYhqd.bDGrAh17tGootLxK.;received=192.168.1.15;rport=5060
From: sip:0260@192.168.0.3;tag=1MHx1fvpNNL5yQngeLpiA-EHIvy9sgEQ
To: sip:0260@192.168.0.3;tag=as1f0a38d2
Call-ID: Yz2wQMhmA6Nnc5MTCkNZIyCHJLI1EvEZ
CSeq: 9604 REGISTER
Server: Asterisk PBX 1.8.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: sip:0260@192.168.1.15:5060;expires=60
Date: Fri, 23 Nov 2012 07:54:14 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘Yz2wQMhmA6Nnc5MTCkNZIyCHJLI1EvEZ’ in 32000 ms (Method: REGISTER)
== Using SIP RTP CoS mark 5
We think we can do text
Audio is at 18682
Adding codec 0x100000000 (g719) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.15:5060:
INVITE sip:0260@192.168.1.15:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK39f6193b;rport
Max-Forwards: 70
From: “0410” sip:0410@192.168.0.3;tag=as77d33fd1
To: sip:0260@192.168.1.15:5060
Contact: sip:0410@192.168.0.3:5060
Call-ID: 0ee1853306ff579d52dc9d0a6558577c@192.168.0.3:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.18.0
Date: Fri, 23 Nov 2012 07:54:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 1443534341 1443534341 IN IP4 192.168.0.3
s=Asterisk PBX 1.8.18.0
c=IN IP4 192.168.0.3
t=0 0
m=audio 18682 RTP/AVP 116 101
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:192.168.1.15:5060 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;received=192.168.0.3;branch=z9hG4bK39f6193b
Call-ID: 0ee1853306ff579d52dc9d0a6558577c@192.168.0.3:5060
From: “0410” sip:0410@192.168.0.3;tag=as77d33fd1
To: sip:0260@192.168.1.15;tag=z9hG4bK39f6193b
CSeq: 102 INVITE
Warning: 399 am180x-evm "No suitable codec for remote offer (PJMEDIA_SDPNEG_NOANSCODEC)"
Accept: application/sdp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Transmitting (NAT) to 192.168.1.15:5060:
ACK sip:0260@192.168.1.15:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK39f6193b;rport
Max-Forwards: 70
From: “0410” sip:0410@192.168.0.3;tag=as77d33fd1
To: sip:0260@192.168.1.15:5060;tag=z9hG4bK39f6193b
Contact: sip:0410@192.168.0.3:5060
Call-ID: 0ee1853306ff579d52dc9d0a6558577c@192.168.0.3:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.18.0
Content-Length: 0

Scheduling destruction of SIP dialog ‘0ee1853306ff579d52dc9d0a6558577c@192.168.0.3:5060’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.15:5060 —>

actually from client pjsua I am using codec g722 and in sip.conf also I have set in general category
disallow=all
allow=g722

please guide me what I am configuring wrong

Thanks in advance
Pranav
a

2

You are requesting G.719 and only G.719. I’d have to assume that you have overridden the codec choice for that peer.

3

Hi david

how can I change it to g722 and from which side client side or asterisk side
I am using pjsua1.8 as client and asterisk 1.8.18.0 I have
pjsua1.8 supports sip version 2.0/udp

Please guide how to negotiate codec g722 as I have set in general category
disallow=all
allow=g722

and

at time of runnig pjsua binary ./pjsua --add-codec=G722 is set

please guide is there any other setting should I do to negotiate codec to g722

Thanks in advance
Pranav

4

What do you have for the peer?

5

hi david

I am using pjsua 1.8 on hard phone based on linux
sip configuration of which is as follows is sip.conf
suppose 0010 is one sip registered from one hard phone

[0011]
username = 0011
secret = 0011
type = friend
host = dynamic
context = users
insecure = port,invite

Please guide me to find cause of this problem

Thanks in advance
Pranav .

6

Please provide the diagnostic output you were asked to provide (the module debugging from chan_sip, which goes into detail as to how it chose the codecs).

Nothing I have seen so far should cause G.719 to be offered. Is anything in the system using G.719?

Incidentally, why do you have insecure set. This is generally not needed for dynamic hosts and, as its name implies, it makes the system less secure. Even when needed, only insecure=invite is usually needed.

7

hi david here is the debug of asterisk you asked

[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:8196 find_call: = Looking for Call ID: rnSbSQCJoXCHVxUnzuwoLQ10Z5D1B…H (Checking From) --From tag CHuMWI8XvOMqKIHu.0Dmsdkb-ssIgznn --To-tag
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:8196 find_call: = Looking for Call ID: rnSbSQCJoXCHVxUnzuwoLQ10Z5D1B…H (Checking From) --From tag CHuMWI8XvOMqKIHu.0Dmsdkb-ssIgznn --To-tag
[Nov 26 01:31:16] DEBUG[2065]: acl.c:736 ast_ouraddrfor: For destination ‘192.168.0.12’, our source address is ‘192.168.0.3’.
[Nov 26 01:31:16] DEBUG[2065]: acl.c:736 ast_ouraddrfor: For destination ‘192.168.0.12’, our source address is ‘192.168.0.3’.
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:3553 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.0.3:5060
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:3553 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.0.3:5060
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:7876 sip_alloc: Allocating new SIP dialog for rnSbSQCJoXCHVxUnzuwoLQ10Z5D1B…H - REGISTER (No RTP)
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:7876 sip_alloc: Allocating new SIP dialog for rnSbSQCJoXCHVxUnzuwoLQ10Z5D1B…H - REGISTER (No RTP)
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:25378 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:25378 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.12:5060’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.12:5060’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.12’ and port ‘5060’.
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.12’ and port ‘5060’.
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.3’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.3’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.3’ and port ‘’.
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.3’ and port ‘’.
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:3401 __sip_xmit: Trying to put ‘SIP/2.0 401’ onto UDP socket destined for 192.168.0.12:5060
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:3401 __sip_xmit: Trying to put ‘SIP/2.0 401’ onto UDP socket destined for 192.168.0.12:5060
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:8196 find_call: = Looking for Call ID: rnSbSQCJoXCHVxUnzuwoLQ10Z5D1B…H (Checking From) --From tag CHuMWI8XvOMqKIHu.0Dmsdkb-ssIgznn --To-tag
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:8196 find_call: = Looking for Call ID: rnSbSQCJoXCHVxUnzuwoLQ10Z5D1B…H (Checking From) --From tag CHuMWI8XvOMqKIHu.0Dmsdkb-ssIgznn --To-tag
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.3’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.3’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.3’ and port ‘’.
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.3’ and port ‘’.
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.3’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.3’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.3’ and port ‘’.
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.3’ and port ‘’.
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:25378 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:25378 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.12:5060’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.12:5060’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.12’ and port ‘5060’.
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.12’ and port ‘5060’.
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.3’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting ‘192.168.0.3’ into…
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.3’ and port ‘’.
[Nov 26 01:31:16] DEBUG[2065]: netsock2.c:192 ast_sockaddr_split_hostport: …host ‘192.168.0.3’ and port ‘’.
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:14243 parse_register_contact: Store REGISTER’s src-IP:port for call routing.
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:14243 parse_register_contact: Store REGISTER’s src-IP:port for call routing.
– Registered SIP ‘0210’ at 192.168.0.12:5060
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:3401 __sip_xmit: Trying to put ‘SIP/2.0 200’ onto UDP socket destined for 192.168.0.12:5060
[Nov 26 01:31:16] DEBUG[2065]: chan_sip.c:3401 __sip_xmit: Trying to put ‘SIP/2.0 200’ onto UDP socket destined for 192.168.0.12:5060
[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/0210
PeerStatus: Registered
Address: 192.168.0.12:5060

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/0210
PeerStatus: Registered
Address: 192.168.0.12:5060

[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 0210
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 0210
[Nov 26 01:31:16] DEBUG[2036]: chan_sip.c:26674 sip_devicestate: Checking device state for peer 0210
[Nov 26 01:31:16] DEBUG[2036]: chan_sip.c:26674 sip_devicestate: Checking device state for peer 0210
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:460 do_state_change: Changing state for SIP/0210 - state 1 (Not in use)
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:460 do_state_change: Changing state for SIP/0210 - state 1 (Not in use)
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:440 devstate_event: device ‘SIP/0210’ state ‘1’
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:440 devstate_event: device ‘SIP/0210’ state ‘1’
[Nov 26 01:31:16] DEBUG[2089]: app_queue.c:1488 handle_statechange: Device ‘SIP/0210’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 26 01:31:16] DEBUG[2089]: app_queue.c:1488 handle_statechange: Device ‘SIP/0210’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 26 01:31:16] DEBUG[2043]: manager.c:4719 process_message: Running action ‘Originate’
[Nov 26 01:31:16] DEBUG[2043]: manager.c:4719 process_message: Running action ‘Originate’
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:26773 sip_request_call: Asked to create a SIP channel with formats: 0x10c7f0000 (jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|g719)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:26773 sip_request_call: Asked to create a SIP channel with formats: 0x10c7f0000 (jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|g719)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7876 sip_alloc: Allocating new SIP dialog for 5f234126776fe42677647df22e93bc73@127.0.0.1:5060 - INVITE (No RTP)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7876 sip_alloc: Allocating new SIP dialog for 5f234126776fe42677647df22e93bc73@127.0.0.1:5060 - INVITE (No RTP)
[Nov 26 01:31:16] DEBUG[2043]: rtp_engine.c:350 ast_rtp_instance_new: Using engine ‘asterisk’ for RTP instance ‘0x90fa588’
[Nov 26 01:31:16] DEBUG[2043]: rtp_engine.c:350 ast_rtp_instance_new: Using engine ‘asterisk’ for RTP instance ‘0x90fa588’
[Nov 26 01:31:16] DEBUG[2043]: res_rtp_asterisk.c:557 ast_rtp_new: Allocated port 14516 for RTP instance ‘0x90fa588’
[Nov 26 01:31:16] DEBUG[2043]: res_rtp_asterisk.c:557 ast_rtp_new: Allocated port 14516 for RTP instance ‘0x90fa588’
[Nov 26 01:31:16] DEBUG[2043]: rtp_engine.c:359 ast_rtp_instance_new: RTP instance ‘0x90fa588’ is setup and ready to go
[Nov 26 01:31:16] DEBUG[2043]: rtp_engine.c:359 ast_rtp_instance_new: RTP instance ‘0x90fa588’ is setup and ready to go
[Nov 26 01:31:16] DEBUG[2043]: res_rtp_asterisk.c:2537 ast_rtp_prop_set: Setup RTCP on RTP instance ‘0x90fa588’
[Nov 26 01:31:16] DEBUG[2043]: res_rtp_asterisk.c:2537 ast_rtp_prop_set: Setup RTCP on RTP instance ‘0x90fa588’
== Using SIP RTP CoS mark 5
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:5160 do_setnat: Setting NAT on RTP to Off
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:5160 do_setnat: Setting NAT on RTP to Off
[Nov 26 01:31:16] DEBUG[2043]: acl.c:736 ast_ouraddrfor: For destination ‘192.168.0.12’, our source address is ‘192.168.0.3’.
[Nov 26 01:31:16] DEBUG[2043]: acl.c:736 ast_ouraddrfor: For destination ‘192.168.0.12’, our source address is ‘192.168.0.3’.
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:3553 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.0.3:5060
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:3553 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.0.3:5060
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7172 sip_new: *** Our native formats are 0x1000 (g722)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7172 sip_new: *** Our native formats are 0x1000 (g722)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7173 sip_new: *** Joint capabilities are 0x0 (nothing)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7173 sip_new: *** Joint capabilities are 0x0 (nothing)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7174 sip_new: *** Our capabilities are 0x1000 (g722)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7174 sip_new: *** Our capabilities are 0x1000 (g722)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7175 sip_new: *** AST_CODEC_CHOOSE formats are 0x1000 (g722)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7175 sip_new: *** AST_CODEC_CHOOSE formats are 0x1000 (g722)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7177 sip_new: *** Our preferred formats from the incoming channel are 0x10c7f0000 (jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|g719)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7177 sip_new: *** Our preferred formats from the incoming channel are 0x10c7f0000 (jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|g719)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7205 sip_new: This channel will not be able to handle video.
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:7205 sip_new: This channel will not be able to handle video.
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:5703 sip_call: Outgoing Call for 0210
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:5703 sip_call: Outgoing Call for 0210
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:5986 update_call_counter: Updating call counter for outgoing call
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:5986 update_call_counter: Updating call counter for outgoing call
[Nov 26 01:31:16] WARNING[2043]: chan_sip.c:5717 sip_call: No audio format found to offer. Cancelling call to 0210
[Nov 26 01:31:16] WARNING[2043]: chan_sip.c:5717 sip_call: No audio format found to offer. Cancelling call to 0210
[Nov 26 01:31:16] NOTICE[2043]: channel.c:5451 __ast_request_and_dial: Unable to call channel SIP/0210
[Nov 26 01:31:16] NOTICE[2043]: channel.c:5451 __ast_request_and_dial: Unable to call channel SIP/0210
[Nov 26 01:31:16] DEBUG[2043]: channel.c:2873 ast_hangup: Hanging up channel ‘SIP/0210-00000003’
[Nov 26 01:31:16] DEBUG[2043]: channel.c:2873 ast_hangup: Hanging up channel ‘SIP/0210-00000003’
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:6368 sip_hangup: Hangup call SIP/0210-00000003, SIP callid 2971c8a26a4ea1d668dc08b83ed7858d@192.168.0.3:5060
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:6368 sip_hangup: Hangup call SIP/0210-00000003, SIP callid 2971c8a26a4ea1d668dc08b83ed7858d@192.168.0.3:5060
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:6387 sip_hangup: Hanging up channel in state Down (not UP)
[Nov 26 01:31:16] DEBUG[2043]: chan_sip.c:6387 sip_hangup: Hanging up channel in state Down (not UP)
[Nov 26 01:31:16] DEBUG[2043]: res_rtp_asterisk.c:2577 ast_rtp_remote_address_set: Setting RTCP address on RTP instance ‘0x90fa588’
[Nov 26 01:31:16] DEBUG[2043]: res_rtp_asterisk.c:2577 ast_rtp_remote_address_set: Setting RTCP address on RTP instance ‘0x90fa588’
[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: Newchannel
Privilege: call,all
Channel: SIP/0210-00000003
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: users
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: Newchannel
Privilege: call,all
Channel: SIP/0210-00000003
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: users
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: SIPCALLID
Value: 2971c8a26a4ea1d668dc08b83ed7858d@192.168.0.3:5060
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: SIPCALLID
Value: 2971c8a26a4ea1d668dc08b83ed7858d@192.168.0.3:5060
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: secret
Value: 0210
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: secret
Value: 0210
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: quality
Value: no
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: quality
Value: no
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: type
Value: friend
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: type
Value: friend
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: host
Value: dynamic
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: VarSet
Privilege: dialplan,all
Channel: SIP/0210-00000003
Variable: host
Value: dynamic
Uniqueid: 1353911476.3

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: NewAccountCode
Privilege: call,all
Channel: SIP/0210-00000003
Uniqueid: 1353911476.3
AccountCode:
OldAccountCode:

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: NewAccountCode
Privilege: call,all
Channel: SIP/0210-00000003
Uniqueid: 1353911476.3
AccountCode:
OldAccountCode:

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: NewCallerid
Privilege: call,all
Channel: SIP/0210-00000003
CallerIDNum: 0410
CallerIDName:
Uniqueid: 1353911476.3
CID-CallingPres: 0 (Presentation Allowed, Not Screened)

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: NewCallerid
Privilege: call,all
Channel: SIP/0210-00000003
CallerIDNum: 0410
CallerIDName:
Uniqueid: 1353911476.3
CID-CallingPres: 0 (Presentation Allowed, Not Screened)

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: Hangup
Privilege: call,all
Channel: SIP/0210-00000003
Uniqueid: 1353911476.3
CallerIDNum: 0410
CallerIDName:
ConnectedLineNum: 0410
ConnectedLineName:
Cause: 0
Cause-txt: Unknown

[Nov 26 01:31:16] DEBUG[2043]: manager.c:4300 match_filter: Examining event:
Event: Hangup
Privilege: call,all
Channel: SIP/0210-00000003
Uniqueid: 1353911476.3
CallerIDNum: 0410
CallerIDName:
ConnectedLineNum: 0410
ConnectedLineName:
Cause: 0
Cause-txt: Unknown

[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 0210
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - 0210
[Nov 26 01:31:16] DEBUG[2036]: chan_sip.c:26674 sip_devicestate: Checking device state for peer 0210
[Nov 26 01:31:16] DEBUG[2036]: chan_sip.c:26674 sip_devicestate: Checking device state for peer 0210
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:460 do_state_change: Changing state for SIP/0210 - state 1 (Not in use)
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:460 do_state_change: Changing state for SIP/0210 - state 1 (Not in use)
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:440 devstate_event: device ‘SIP/0210’ state ‘1’
[Nov 26 01:31:16] DEBUG[2036]: devicestate.c:440 devstate_event: device ‘SIP/0210’ state ‘1’
[Nov 26 01:31:16] DEBUG[2089]: app_queue.c:1488 handle_statechange: Device ‘SIP/0210’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 26 01:31:16] DEBUG[2089]: app_queue.c:1488 handle_statechange: Device ‘SIP/0210’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 26 01:31:16] DEBUG[2043]: manager.c:4719 process_message: Running action ‘Getvar’
[Nov 26 01:31:16] DEBUG[2043]: manager.c:4719 process_message: Running action ‘Getvar’
[Nov 26 01:31:18] DEBUG[2065]: chan_sip.c:3941 __sip_autodestruct: Auto destroying SIP dialog ‘nw1Fzy8owpCmiyIrWMUDXRegwvDyVK-M’
[Nov 26 01:31:18] DEBUG[2065]: chan_sip.c:3941 __sip_autodestruct: Auto destroying SIP dialog ‘nw1Fzy8owpCmiyIrWMUDXRegwvDyVK-M’
[Nov 26 01:31:18] DEBUG[2065]: chan_sip.c:6134 sip_destroy: Destroying SIP dialog nw1Fzy8owpCmiyIrWMUDXRegwvDyVK-M
[Nov 26 01:31:18] DEBUG[2065]: chan_sip.c:6134 sip_destroy: Destroying SIP dialog nw1Fzy8owpCmiyIrWMUDXRegwvDyVK-M
[Nov 26 01:31:26] DEBUG[2065]: chan_sip.c:3941 __sip_autodestruct: Auto destroying SIP dialog ‘up4P6X.ja6e7eLNrTcaCSXlDRIo4JRNh’
[Nov 26 01:31:26] DEBUG[2065]: chan_sip.c:3941 __sip_autodestruct: Auto destroying SIP dialog ‘up4P6X.ja6e7eLNrTcaCSXlDRIo4JRNh’
[Nov 26 01:31:26] DEBUG[2065]: chan_sip.c:6134 sip_destroy: Destroying SIP dialog up4P6X.ja6e7eLNrTcaCSXlDRIo4JRNh
[Nov 26 01:31:26] DEBUG[2065]: chan_sip.c:6134 sip_destroy: Destroying SIP dialog up4P6X.ja6e7eLNrTcaCSXlDRIo4JRNh

Please help me to find cause of this problem

Thanks in advance
Pranav

8

Is there a good reason why you are not using g711 codec?
The g711a/u should give you very little grief, because it is supported by every major VoIP provider and IP Phone / Gateway manufacturer. When you go and use a high-compression codec, you usually end up with lower voice quality and IOP issues.

9

I am surprised that it has offered G.719, in spite of the restrictions in sip.conf. However, a key fact that was missed from the original problem statement is that this is an AMI Originate. When you originate a call, you constrain the codecs that are used internally.

Do you actually have G.722 codecs, or are you operating in pass through mode? Whilst there may have been some changes since I looked at the code in detail, if you want to do pass through with Originate, you must Originate the call with the codec that you intend to pass through.

What is strange, and may reflect a change since I looked at it, is that the originate seems to be requesting G.719, when it, at least, used to default to signed linear. I’m wondering if your AMI is explicitly requesting G.719.

To add to what dejanst says, if you use the right variant of G.711, you will get no additional degradation from the codec, as these are the codecs used by the PSTN (the choice between A- and mu- depends on where you are in the world).

10

Thanks a lot David

actually though I was not setting any codec in AMI originateAcation the codec was automatically assigned to g719 but now I have set it to g722 explicitly so it is working

Thanks & Regards
Pranav