* You are viewing Posts Tagged ‘rtpmap’

No progress tones on transferred call

Asterisk 1.4
We are experiencing an issue on transfers where no progress tones are heard
by the caller:
1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF).
1595 answers
2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears
MoH.
3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595
hears no ringing
When xfer is pressed and the extension is dialled:
U 203.89.001.001:5060 -> 121.98.001.001:1034
INVITE sip:1CDF0F4AFFFF@192.168.1.72:5060 SIP/2.0..Via: SIP/2.0/UDP
203.89.001.001:5060;branch=z9hG4bK5286810e;rport..From: “C Allerid”
;tag=as72616c50..To:
..Contact:
..Call-ID:
59ba10300b9b8cb5684eba2368c90a77@203.89.001.001..CSeq: 102
INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 05 Jun 2012
08:05:02 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO..Supported: replaces..Content-Type:
application/sdp..Content-Length: 262….v=0..o=root 3031 3031 IN IP4
203.89.001.001..s=session..c=IN IP4 203.89.001.001..t=0 0..m=audio 13728
RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101
telephone-event/8000..a=fmtp:1010-16..a=ptime:20..a=sendrecv..
U 121.98.001.001:1034 -> 203.89.001.001:5060
SIP/2.0 100 Trying..To: ..From: “C
Allerid” ;tag=as72616c50..Call-ID:
59ba10300b9b8cb5684eba2368c90a77@203.89.001.001..CSeq: 102 INVITE..Via:
SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Server:
Cisco/SPA508G-7.4.9c..Content-Length: 0….
U 121.98.001.001:1034 -> 203.89.001.001:5060
SIP/2.0 180 Ringing..To:
;tag=53e23c5265d60f06i0..From: “C
Allerid”
;tag=as72616c50..Call-ID:59ba10300b9b8cb5684eba2368
c90a77@203.89.001.001..CSeq: 102 INVITE..Via: SIP/2.0/UDP
203.89.001.001:5060;branch=z9hG4bK5286810e..Contact: “$USER”
..Server:
Cisco/SPA508G-7.4.9c..Content-Length: 0….
After transfer is pressed the second time there is no further SIP messages
with

Asterisk CLI

Error about codecs AMR-NB.

 Hi.
 Anyone know how to fix this problem below.
 I’m add codecs AMR-NB and AMR-WB and I get error in AMR-NB about this “Found unknown media description format AMR for ID”, a search about this on google and I can’t find any solution about this. Thanks in advanced and best regards.

             Julio Lemos

< ------------->— (8 headers 0 lines) —
< --- SIP read from UDP:192.168.3.227:45327 --->INVITE sip:5432@192.168.3.148 SIP/2.0Via: SIP/2.0/UDP 192.168.3.227:45327;rport;branch=z9hG4bKPjQv3ZDrKzwWFMfOO4eFG0AECvwqEerQ9AMax-Forwards: 70From: “5505″ ;tag=H24FA8xsH7K9DpuES9CbWgGkvbYSfF8qTo: Contact: “5505″ Call-ID: WyL8a33LsNOaQjw2WbVUN7qj0R4hyEMOCSeq: 13600 INVITEAllow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONSSupported: replaces, 100rel, timer, norefersubSession-Expires: 1800Min-SE: 90User-Agent: CSipSimple r1108 / umts_milestone2-10Authorization: Digest username=”5505″, realm=”asterisk”, nonce=”52b8e2d3″, uri=”sip:5432@192.168.3.148″, response=”1a95c0bbb6f5143037eb6f7b6f2f6674″, algorithm=MD5Content-Type: application/sdpContent-Length: 305
v=0o=- 3547457675 3547457675 IN IP4 192.168.3.227s=pjmediac=IN IP4 192.168.3.227t=0 0a=X-nat:0m=audio 4000 RTP/AVP 108 8 0 101a=rtcp:4001 IN IP4 192.168.3.227a=rtpmap:108 AMR/8000a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=sendrecva=rtpmap:101 telephone-event/8000a=fmtp:101 0-15< ------------->— (16 headers 14 lines) —Sending to 192.168.3.227:45327 (NAT)Using INVITE request as basis request – WyL8a33LsNOaQjw2WbVUN7qj0R4hyEMOFound peer ’5505′ for ’5505′ from 192.168.3.227:45327  == Using SIP RTP CoS mark 5Found RTP audio format 108Found RTP audio format 8Found RTP audio format 0Found RTP audio format 101Found unknown media description format AMR for ID 108Found audio description format PCMA for ID 8Found audio description format PCMU for ID 0Found audio description format telephone-event for ID 101[May 31 09:55:05] NOTICE[28286]: chan_sip.c:9372 process_sdp: No compatible codecs, not accepting this offer!
< --- Reliably Transmitting (NAT) to 192.168.3.227:45327 --->SIP/2.0 488 Not acceptable hereVia: SIP/2.0/UDP 192.168.3.227:45327;branch=z9hG4bKPjQv3ZDrKzwWFMfOO4eFG0AECvwqEerQ9A;received=192.168.3.227;rport=45327From: “5505″
;tag=H24FA8xsH7K9DpuES9CbWgGkvbYSfF8qTo: ;tag=as02c2bc10Call-ID: WyL8a33LsNOaQjw2WbVUN7qj0R4hyEMOCSeq: 13600 INVITEServer: Asterisk PBX 10.4.0Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContent-Length: 0

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

Can’t make Asterisk send authentication to remote peer on INVITE

This is a really simple problem that I just can’t get to work. There
are two Asterisk servers with the following sip user and peer. When a
call is attempted, Asterisk is not sending authentication details in
response to the 401. Note, if the secret is blank on 172.16.0.2 test,
the INVITE succeeds.

on 172.16.0.2:

[test]
type=friend
secret=abcde
host=dynamic
context=demo

on 172.16.0.1 :

[natty]
type=peer
host=172.16.0.2
fromuser=test
secret=abcde

originate SIP/natty/1234568 extension 200
== Using SIP RTP CoS mark 5
Audio is at 172.16.0.1 port 19486
Adding codec 0×2 (gsm) to SDP
Adding codec 0×4 (ulaw) to SDP
Adding codec 0×8 (alaw) to SDP
Adding non-codec 0×1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.0.2:5060:
INVITE sip:1234568@172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
Max-Forwards: 70
From: “asterisk” ;tag=as1689b2b6
To:
Contact:
Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Date: Sat, 14 Apr 2012 09:10:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1594270426 1594270426 IN IP4 172.16.0.1
s=Asterisk PBX 1.6.2.9-2ubuntu2.1
c=IN IP4 172.16.0.1
t=0 0
m=audio 19486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

SIP trunk call initiated as Anonymous@anonymous.invalid

I have a Grandstream HT-502 device connected to my Asterisk PBX. It is
configured not to place anonymous calls, and from my mostly layman
reading of the invitation that the device sends, it should not be
anonymous. However, the Asterisk PBX sends an anonymous invitation to
our SIP trunk provider. Can anyone explain why?

The two INVITE packets follow.

The devices sends the following INVITE:

INVITE sip:2223334444@pbx.xxxxx.com SIP/2.0
Via: SIP/2.0/UDP 192.168.9.197:46538;branch=z9hG4bK526774101;rport
From: “222333555″ ;tag=2072922124
To:
Call-ID: 1082640776-46538-3@BJC.BGI.J.BJH
CSeq: 21 INVITE
Contact: “222333555″
Authorization: Digest username=”222333555″, realm=”asterisk”,
nonce=”02774xxx”, uri=”sip:2223334444@pbx.xxxxx.com”,
response=”0d1b93729332670aae5b6916ecfxxxxx”, algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-502 V1.2A 1.0.5.10
Privacy: none
P-Asserted-Identity: “222333555″

Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 400

v=0
o=222333555 8000 8000 IN IP4 192.168.9.197
s=SIP Call
c=IN IP4 192.168.9.197
t=0 0
m=audio 58270 RTP/AVP 0 8 4 18 112 97 102 100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000

Our PBX sends this INVITE to our SIP trunk provider:

INVITE sip:2223334444@10.250.0.5 SIP/2.0
Via: SIP/2.0/UDP 66.77.88.99:5060;branch=z9hG4bK1b55d480;rport
Max-Forwards: 70
From: “Anonymous” ;tag=as567ac377
To:
Contact:
Call-ID: 08be883c133cae41515d1f914d62f6ce@66.77.88.99:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.7.2)
Date: Thu, 12 Jan 2012 19:55:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 525025075 525025075 IN IP4 66.77.88.99
s=Asterisk PBX 1.8.7.2
c=IN IP4 66.77.88.99
t=0 0
m=audio 15408 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

No rtpmap codec info in 200 OK

Hi,

My VSP uses Asterisk to which I’m connected with an ATA.

When I receive an inbound call the invite includes the following…

v=0
o=root 32218 32218 IN IP4 202.52.129.50
s=session
c=IN IP4 202.52.129.50
t=0 0
m=audio 16864 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off – – – -
a=ptime:20
a=sendrecv

My ATA’s 200 OK reply after call setup has the following…

v=0
o=CMI-SIPUA 13369 0 IN IP4 211.30.XXX.XXX
s=SIP CALL
c=IN IP4 211.30.XXX.XXX
t=0 0
m=audio 20216 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=rtcp:20217
a=silenceSupp:off – – – -
a=sendrecv

Notice there is no “rtpmap:18 G729/8000″ in the reply.

The call continues fine.

Is it right that there is no codec info in the reply and the call continues?

a=sendonly Music On Hold ignored

Hello all,

I have a problem of “Music on Hold” on AsteriskNow system, based on Asterisk
1.6.2.19 with FreePBX 2.8.1.4

On another system, when we press the HOLD button on the phone, the phone
sends an INVITE with a=sendonly in the SDP, and we get an OK and the system
recognizes the a=sendonly request and starts the music on hold, as you can
see from the following log:

< --- SIP read from UDP:10.0.0.2:5060 --->
INVITE sip:21@10.0.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;rport;branch=z9hG4bK337477455
From: “1001″ ;tag=446928907
To: ;tag=as479a82ac
Call-ID: 65159842@10.0.0.2
CSeq: 22 INVITE
Contact:
Max-Forwards: 70
User-Agent: sip phone
Subject: Phone call
Content-Type: application/sdp
Content-Length: 419

v=0
o=1001 0000000001 0000000002 IN IP4 10.0.0.2
s=A conversation
c=IN IP4 0.0.0.0
t=0 0
m=audio 9000 RTP/AVP 18 4 0 8 23 22 2 21 3 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:23 G726-16/8000
a=rtpmap:22 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:21 G726-40/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly

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