No progress tones on transferred call

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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:

Asterisk Users 3.1 years ago 0 Answer

Error about codecs AMR-NB.

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 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,…

Asterisk Users 3.1 years ago 0 Answer

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

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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

Asterisk Users 3.3 years ago 2 Answer

SIP trunk call initiated as Anonymous@anonymous.invalid

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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:…

Asterisk Users 3.5 years ago 0 Answer

No rtpmap codec info in 200 OK

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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…

Asterisk Users 3.6 years ago 7 Answer