Hello,
I am trying to setup a Cisco SPA504G to run on asterisk. Everything works but the ability to dial out. I also have 2 snom phones connected which both work perfectly. Yet another sipura phone shares the same problem that the SPA504G has. I get a message on screen “NotAcceptableHere” along with a very fast busy tone. Thank you in advance for your time.
Steve
Hello. Try to verify you login/password on phone. My be somthing wrong. Make command “sip show peers”. If you can do internal call between 2 SNOM’s phones, my be trouble in yours SIP trunk. I use this phone many time and I don’t have trouble with him.
Hi,
Check out this page supportforums.cisco.com/docs/DOC-9955 for configuring SPA5xx phones with asterisk.
Stoyan
Thank you for the replies. The problem persists. The SNOM phones can dial out to any number and receive calls. The SIPURA and CISCO cannot dial out anywhere but can receive calls. I tried following the steps from that manual provided by SGM but no success. Still looking for a solution. Please help 
Turn on sip debugging with sip set debug on
Then copy and paste everything from the incoming INVITE through to the not allowed message, here. Remove any sensitive data.
This is the log:
<------------->
[Aug 24 12:55:24] VERBOSE[1891] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
NOTIFY sip:222.222.222.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK-fc6f46c8
From: “Asterisk” sip:9876@222.222.222.222;tag=bbf7941ac34031a3o0
To: sip:222.222.222.222
Call-ID: 3232d9c6-817fc7e7@192.168.1.116
CSeq: 38 NOTIFY
Max-Forwards: 70
Contact: “Asterisk” sip:9876@192.168.1.116:5060
Event: keep-alive
User-Agent: Cisco/SPA504G-7.4.8
Content-Length: 0
<------------->
[Aug 24 12:55:24] VERBOSE[1891] chan_sip.c: — (11 headers 0 lines) —
[Aug 24 12:55:24] VERBOSE[1891] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK-fc6f46c8;received=192.168.1.1
From: “Asterisk” sip:9876@222.222.222.222;tag=bbf7941ac34031a3o0
To: sip:222.222.222.222;tag=as3a970889
Call-ID: 3232d9c6-817fc7e7@192.168.1.116
CSeq: 38 NOTIFY
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[Aug 24 12:55:24] VERBOSE[1891] chan_sip.c: Scheduling destruction of SIP dialog ‘3232d9c6-817fc7e7@192.168.1.116’ in 32000 ms (Method: NOTIFY)
[Aug 24 12:55:24] VERBOSE[1891] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:2051:
OPTIONS sip:1234@192.168.1.11:2051;line=mqqljl6v SIP/2.0
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK19e72b46;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@222.222.222.222;tag=as4996595b
To: sip:1234@192.168.1.11:2051;line=mqqljl6v
Contact: sip:Unknown@222.222.222.222:5060
Call-ID: 3920aa692769ae5258a006280a147f18@222.222.222.222:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.3.2)
Date: Wed, 24 Aug 2011 16:55:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
[Aug 24 12:55:25] VERBOSE[1891] chan_sip.c:
<— SIP read from UDP:192.168.1.1:2051 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK19e72b46;rport=5060
From: “Unknown” sip:Unknown@222.222.222.222;tag=as4996595b
To: sip:1234@192.168.1.11:2051;line=mqqljl6v
Call-ID: 3920aa692769ae5258a006280a147f18@222.222.222.222:5060
CSeq: 102 OPTIONS
Contact: sip:1234@192.168.1.11:2051;line=mqqljl6v;flow-id=1
User-Agent: snom300/6.5.20
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Content-Length: 0
[Aug 24 12:55:25] VERBOSE[1891] chan_sip.c: Really destroying SIP dialog ‘5563edd90f8d016664fb008e1fed7a68@222.222.222.222:5060’ Method: OPTIONS
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
INVITE sip:1234@222.222.222.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK-58502058
From: “Asterisk” sip:9876@222.222.222.222;tag=dc23427518584150o0
To: “Test Line” sip:1234@222.222.222.222
Call-ID: f37e5119-acf5e2fc@192.168.1.116
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “Asterisk” sip:9876@192.168.1.116:5060
Expires: 240
User-Agent: Cisco/SPA504G-7.4.8
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 63170 63170 IN IP4 192.168.1.116
s=-
c=IN IP4 192.168.1.116
t=0 0
m=audio 16458 RTP/AVP 9 0 2 8 18 96 97 98 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: — (14 headers 18 lines) —
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Sending to 192.168.1.1:5060 (no NAT)
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Using INVITE request as basis request - f37e5119-acf5e2fc@192.168.1.116
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found peer ‘9876’ for ‘9876’ from 192.168.1.1:5060
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c:
<— Reliably Transmitting (NAT) to 192.168.1.1:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK-58502058;received=192.168.1.1;rport=5060
From: “Asterisk” sip:9876@222.222.222.222;tag=dc23427518584150o0
To: “Test Line” sip:1234@222.222.222.222;tag=as0406c575
Call-ID: f37e5119-acf5e2fc@192.168.1.116
CSeq: 101 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0060928b"
Content-Length: 0
<------------>
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Scheduling destruction of SIP dialog ‘f37e5119-acf5e2fc@192.168.1.116’ in 6400 ms (Method: INVITE)
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
ACK sip:1234@222.222.222.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK-58502058
From: “Asterisk” sip:9876@222.222.222.222;tag=dc23427518584150o0
To: “Test Line” sip:1234@222.222.222.222;tag=as0406c575
Call-ID: f37e5119-acf5e2fc@192.168.1.116
CSeq: 101 ACK
Max-Forwards: 70
Contact: “Asterisk” sip:9876@192.168.1.116:5060
User-Agent: Cisco/SPA504G-7.4.8
Content-Length: 0
<------------->
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: — (10 headers 0 lines) —
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
INVITE sip:1234@222.222.222.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK-ee3db2f4
From: “Asterisk” sip:9876@222.222.222.222;tag=dc23427518584150o0
To: “Test Line” sip:1234@222.222.222.222
Call-ID: f37e5119-acf5e2fc@192.168.1.116
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“9876”,realm=“asterisk”,nonce=“0060928b”,uri="sip:1234@222.222.222.222",algorithm=MD5,response="a49aa901643496c7a837957fe6cfe5ce"
Contact: “Asterisk” sip:9876@192.168.1.116:5060
Expires: 240
User-Agent: Cisco/SPA504G-7.4.8
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 63170 63170 IN IP4 192.168.1.116
s=-
c=IN IP4 192.168.1.116
t=0 0
m=audio 16458 RTP/AVP 9 0 2 8 18 96 97 98 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: — (15 headers 18 lines) —
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Sending to 192.168.1.1:5060 (NAT)
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Using INVITE request as basis request - f37e5119-acf5e2fc@192.168.1.116
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found peer ‘9876’ for ‘9876’ from 192.168.1.1:5060
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found RTP audio format 9
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found RTP audio format 0
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found RTP audio format 2
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found RTP audio format 8
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found RTP audio format 18
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found RTP audio format 96
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found RTP audio format 97
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found RTP audio format 98
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found RTP audio format 101
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found audio description format G722 for ID 9
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found audio description format PCMU for ID 0
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found audio description format G726-32 for ID 2
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found audio description format PCMA for ID 8
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found audio description format G729a for ID 18
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found audio description format G726-40 for ID 96
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found audio description format G726-24 for ID 97
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found audio description format G726-16 for ID 98
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Found audio description format telephone-event for ID 101
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Capabilities: us - 0x3c050f (g723|gsm|ulaw|alaw|g729|ilbc|h261|h263|h263p|h264), peer - audio=0x101d0c (ulaw|alaw|g726|g729|ilbc|g722|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10050c (ulaw|alaw|g729|ilbc|h263p)
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Peer audio RTP is at port 192.168.1.116:16458
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Peer doesn’t provide video
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Looking for 1234 in from-internal (domain 222.222.222.222)
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: list_route: hop: sip:9876@192.168.1.116:5060
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c:
<— Transmitting (NAT) to 192.168.1.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK-ee3db2f4;received=192.168.1.1;rport=5060
From: “Asterisk” sip:9876@222.222.222.222;tag=dc23427518584150o0
To: “Test Line” sip:1234@222.222.222.222
Call-ID: f37e5119-acf5e2fc@192.168.1.116
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1234@222.222.222.222:5060
Content-Length: 0
<------------>
[Aug 24 12:55:30] VERBOSE[8073] chan_sip.c: Audio is at 5060
[Aug 24 12:55:30] VERBOSE[8073] chan_sip.c: Video is at 222.222.222.222:5060
[Aug 24 12:55:30] VERBOSE[8073] chan_sip.c: Adding codec 0x400 (ilbc) to SDP
[Aug 24 12:55:30] VERBOSE[8073] chan_sip.c: Adding video codec 0x100000 (h263p) to SDP
[Aug 24 12:55:30] VERBOSE[8073] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Aug 24 12:55:30] VERBOSE[8073] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:2051:
INVITE sip:1234@192.168.1.11:2051;line=mqqljl6v SIP/2.0
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK3b8a28bf;rport
Max-Forwards: 70
From: “Volkan Rivera” sip:9876@222.222.222.222;tag=as3cfa5604
To: sip:1234@192.168.1.11:2051;line=mqqljl6v
Contact: sip:9876@222.222.222.222:5060
Call-ID: 2d47786b3e56992f3cccc4721f1a16d3@222.222.222.222:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.3.2)
Date: Wed, 24 Aug 2011 16:55:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 363
v=0
o=root 1766655661 1766655661 IN IP4 222.222.222.222
s=Asterisk PBX 1.8.3.2
c=IN IP4 222.222.222.222
b=CT:384
t=0 0
m=audio 17698 RTP/AVP 97 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:30
a=sendrecv
m=video 11490 RTP/AVP 98
a=rtpmap:98 h263-1998/90000
a=sendrecv
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c:
<— SIP read from UDP:192.168.1.1:2051 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK3b8a28bf;rport=5060
From: “Volkan Rivera” sip:9876@222.222.222.222;tag=as3cfa5604
To: sip:1234@192.168.1.11:2051;line=mqqljl6v;tag=cd6zwbsnq6
Call-ID: 2d47786b3e56992f3cccc4721f1a16d3@222.222.222.222:5060
CSeq: 102 INVITE
Contact: sip:1234@192.168.1.11:2051;line=mqqljl6v;flow-id=1
Warning: 304 x-snom-adr "No supported media type found"
Content-Length: 0
<------------->
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: — (9 headers 0 lines) —
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Transmitting (NAT) to 192.168.1.1:2051:
ACK sip:1234@192.168.1.11:2051;line=mqqljl6v SIP/2.0
Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK3b8a28bf;rport
Max-Forwards: 70
From: “Volkan Rivera” sip:9876@222.222.222.222;tag=as3cfa5604
To: sip:1234@192.168.1.11:2051;line=mqqljl6v;tag=cd6zwbsnq6
Contact: sip:9876@222.222.222.222:5060
Call-ID: 2d47786b3e56992f3cccc4721f1a16d3@222.222.222.222:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.3.2)
Content-Length: 0
[Aug 24 12:55:30] WARNING[8073] channel.c: Unable to find a codec translation path from 0x100400 (ilbc|h263p) to 0x40 (slin)
[Aug 24 12:55:30] WARNING[8073] indications.c: Unable to set ‘SIP/9876-000001fc’ to signed linear format (write)
[Aug 24 12:55:30] NOTICE[8073] app_playtones.c: Unable to start playtones
[Aug 24 12:55:30] VERBOSE[8073] chan_sip.c: Scheduling destruction of SIP dialog ‘f37e5119-acf5e2fc@192.168.1.116’ in 6400 ms (Method: INVITE)
[Aug 24 12:55:30] VERBOSE[8073] chan_sip.c:
<— Reliably Transmitting (NAT) to 192.168.1.1:5060 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK-ee3db2f4;received=192.168.1.1;rport=5060
From: “Asterisk” sip:9876@222.222.222.222;tag=dc23427518584150o0
To: “Test Line” sip:1234@222.222.222.222;tag=as16f8791b
Call-ID: f37e5119-acf5e2fc@192.168.1.116
CSeq: 102 INVITE
Server: FPBX-2.8.1(1.8.3.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c: Really destroying SIP dialog ‘2d47786b3e56992f3cccc4721f1a16d3@222.222.222.222:5060’ Method: INVITE
[Aug 24 12:55:30] VERBOSE[1891] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
ACK sip:1234@222.222.222.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116:5060;branch=z9hG4bK-ee3db2f4
From: “Asterisk” sip:9876@222.222.222.222;tag=dc23427518584150o0
To: “Test Line” sip:1234@222.222.222.222;tag=as16f8791b
Call-ID: f37e5119-acf5e2fc@192.168.1.116
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“9876”,realm=“asterisk”,nonce=“0060928b”,uri="sip:1234@222.222.222.222",algorithm=MD5,response="a49aa901643496c7a837957fe6cfe5ce"
Contact: “Asterisk” sip:9876@192.168.1.116:5060
User-Agent: Cisco/SPA504G-7.4.8
Content-Length: 0
The SNOM doesn’t have an ILBC codec.
It seems you have a codec negotiation issue. As far as I understand one of the phones is using iLBC and the other one is using slin:
[Aug 24 12:55:30] WARNING[8073] channel.c: Unable to find a codec translation path from 0x100400 (ilbc|h263p) to 0x40 (slin)
Check your codec settings and you should be fine.
P. S. You could check what asterisk could translate by running “core show translation”. “-” means it can’t translate this.
slin (16 bit signed linear) is the Asterisk native internal format. It is unlikely that a phone would support it and I’m not even sure that there would be a way for a phone to negotiate it. The conversion is required here so that Asterisk can send a busy tone back to the caller. For some reason, iLBC seems to be the only acceptable codec in common between the caller and what Asterisk thinks the callee can do, but the callee can’t actually do it. Also Asterisk doesn’t have an iLBC codec, so it cannot convert iLBC to something that might be acceptable, or generate tones in iLBC.
I think we need to see sip.conf to find out why iLBC is the only offered codec outbound.
[quote=“SGM”]It seems you have a codec negotiation issue. As far as I understand one of the phones is using iLBC and the other one is using slin:
[Aug 24 12:55:30] WARNING[8073] channel.c: Unable to find a codec translation path from 0x100400 (ilbc|h263p) to 0x40 (slin)
Check your codec settings and you should be fine.
P. S. You could check what asterisk could translate by running “core show translation”. “-” means it can’t translate this.[/quote]
David55, in his follow-on, is correct. It’s not that either phone wants slin. It’s that the first phone is capable of a number of codecs (combined - 0x10050c (ulaw|alaw|g729|ilbc|h263p)) and the second phone is only invited with iLBC:
m=audio 17698 RTP/AVP 97 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
…probably because the second phone’s peer definition is like:
disallow=all
allow=ilbc
where the second phone doesn’t actually support that media type.
Here is the sip.conf. Thank you so much for the help.
[1234]
deny=0.0.0.0/0.0.0.0
secret=zh55555
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/1234
mailbox=1234@device
permit=0.0.0.0/0.0.0.0
callerid=device <1234>
callcounter=yes
faxdetect=no
[4444]
deny=0.0.0.0/0.0.0.0
secret=zh44444
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=no
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/4444
mailbox=4444@device
permit=0.0.0.0/0.0.0.0
callerid=device <4444>
callcounter=yes
faxdetect=no
I agree, didn’t pay too much attention to the entire sip debug, just noticed it’s a codec negotiation issue.
This configuration doesn’t seem to either deny or permit any specific codec. You should check the general configuration of sip.conf and the phone configurations, although, from the debug it seems the phone offers a variety of codecs. Try enabling ulaw in the phones and asterisk itself, at least for testing, or try installing ilbc codec in asterisk.
Thank you so much! Installing the ilbc codec in asterisk solved the problem. 
It’s more like a workaround, but if you’re happy with it 