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).
2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears
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"
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:firstname.lastname@example.org SIP/2.0Via: SIP/2.0/UDP 192.168.3.227:45327;rport;branch=z9hG4bKPjQv3ZDrKzwWFMfOO4eFG0AECvwqEerQ9AMax-Forwards: 70From: "5505"
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]
context=demo on 172.16.0.1 : [natty]
secret=abcde originate SIP/natty/1234568 extension 200
== Using SIP RTP CoS mark 5
Audio is at 172.16.0.1 port 19486
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:email@example.com SIP/2.0
Via: SIP/2.0/UDP 192.168.9.197:46538;branch=z9hG4bK526774101;rport
CSeq: 21 INVITE
My VSP uses Asterisk to which I'm connected with an ATA.
When I receive an inbound call the invite includes the following...
o=root 32218 32218 IN IP4 184.108.40.206
c=IN IP4 220.127.116.11
m=audio 16864 RTP/AVP 18 101
a=silenceSupp:off – - – -
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
c=IN IP4 211.30.XXX.XXX