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Summary:ASTERISK-02177: No Ringback for ingress PSTN calls
Reporter:bcoppens (bcoppens)Labels:
Date Opened:2004-08-02 12:07:35Date Closed:2004-09-25 02:11:25
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) SIP_Asterisk.zip
Description:Calling party does not recieve any ringback when calling SIP to SIP through the asterisk server. Dial plan includes the r attribute and the progressinband is set to yes. The Asterisk pbx does not send any RTP streams after the SIP-180 call state. When answering, we get both way speech( see trace).

FYI:Veraz does not support re-invite. (canreinvite is set to no)

This call setup phenomen is only applicable for the following scenario:

Call setup direction: PSTN- Veraz Class 4(MGCP) - VERAZ NGN(SIP)- ASTERISK(SIP)-Grandstream BT100(SIP)
No nat, all static routes

All other possible scenario's work just fine!

Thx for your assistance!






****** ADDITIONAL INFORMATION ******

Veraz info

Media GW Igate (10.x.x.x)
Control switch MGCP/SIP(172.x.x.100)

Grandstream Ip phone (172.x.x.127)

Asterisk (172.x.x.120)
Asterisk CVS-HEAD-07/27/04-11:27:52 built by root@CCASoftPBX on a i686 running Linux
Linux Redhat version 9


sip.conf

[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
disallow=all
allow=ulaw
progressinband=yes      

[3271256307]
type=friend
context=default
host=172.16.100.100

[6003]
type=friend
context=default
username=6003
secret=
callerid=Dean Crawford <6002>
host=dynamic

Please find extensions.conf enclosed
Comments:By: bcoppens (bcoppens) 2004-08-03 07:10:18

Did some SIP debugging and noticed that the destionation port number is not parsed correctly. ported is set to "0"

See log below

Sip read:
INVITE sip:6003@172.16.100.120:5060 SIP/2.0
Call-ID: 7003097420561121677-1091533775@172.16.100.100
From: sip:3271256307@172.16.100.100:5060;tag=15466
To: sip:6003@172.16.100.120:5060
Content-Length: 273
Content-Type: application/sdp
CSeq: 1 INVITE
Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-613000000000018d-ac106464
Contact: sip:3271256307@172.16.100.100:5060
Max-Forwards: 70

v=0
o=MG4000|1.0 111 12345 IN IP4 10.100.1.21
s=-
c=IN IP4 10.100.1.21
t=0 0
m=audio 58664 RTP/AVP 100 18 0 103
a=rtpmap:100 G.729b/8000
a=rtpmap:103 telephone-event/8000
a=fmtp:103 0-15
a=X-sqn: 0
a=X-cap: 1 image udptl t38
a=ptime:40
m=image 58664 udptl t38

10 headers, 13 lines
Using latest request as basis request
Sending to 172.16.100.100 : 5060 (non-NAT)
Found RTP audio format 100
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 103
Peer RTP is at port 10.100.1.21:0
Found description format G.729b
Found description format telephone-event
Capabilities: us - 0x4(ULAW), peer - audio=0x104(ULAW|G729A)/video=0x0(EMPTY), c
ombined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found user '3271256307'
Looking for 6003 in default
list_route: hop: <sip:3271256307@172.16.100.100:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.100.100:5060;branch=z9hG4bK-613000000000018d-ac106464
From: sip:3271256307@172.16.100.100:5060;tag=15466
To: sip:6003@172.16.100.120:5060;tag=as07924fb0
Call-ID: 7003097420561121677-1091533775@172.16.100.100
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6003@172.16.100.120>
Content-Length: 0
By: bcoppens (bcoppens) 2004-08-05 07:16:22

Reminder sent to markster

Hi Mark,

A fix for this one could tribute to a tremendous achievement for me. Is there anything I can do/check to solve this one?

Thanks,
Bart
By: Mark Spencer (markster) 2004-08-05 18:32:01

Have you purchased a G.729 license?
By: Mark Spencer (markster) 2004-08-05 18:34:08

If you're in a big hurry, you might purchase some time to get this resolved through Digium technical support rather than wait on whenever i'm going to have time on the bug tracker.
By: bcoppens (bcoppens) 2004-08-06 08:08:42

We have purchased a G729 license but only G711ulaw is allowed on my asterisk server. I will try to disable T38 on the Veraz side as I think that maybe the "m=image 58664 udptl t38" interferes.

If this does not work, I will drop the mission,....so you will be able to close the case. Too bad
By: Mark Spencer (markster) 2004-08-06 09:09:12

The invite you posted clearly indicates only G.729 is permitted in the call.
By: bcoppens (bcoppens) 2004-08-06 09:45:42

I think I found the reason; the fact is that Asterisk is using the timing of the input stream to send the output stream. If the Veraz is set to recvonly, no ringback can be genareted from the asterisk. Is there any workarround for this?
By: bcoppens (bcoppens) 2004-08-06 09:48:00

To answer on your remark Mark, the invite is requesting ulaw and G729
By: Mark Spencer (markster) 2004-08-06 10:03:06

You're right, it does have ulaw, it just seems to omit it from the rtpmap for some reason.

Also, you're right, Asterisk synchronizes to the received audio for running generators.  This causes the transmitted audio to be phase-locked to the received audio and thus have no potential for slip.  However if you're not sending audio it won't be received.

It would be possible to use a zap timer should the received audio not be received but that would take some development.

In any case, though, you may run into other problems if the CPE does not send audio in response to our 183 Progress because certain services (e.g. FedEx) require the receipt of early audio (DTMF in this case, specifically) from the phone even before the call is answered.
By: bcoppens (bcoppens) 2004-08-06 11:23:04

But a lot of MGCP integrators only propose send/recv after the Answer supervision. So in all these cases you won't be able to deliver ringback. I will check this behavior in the ITU spec's. I will come back on this subject.
By: Mark Spencer (markster) 2004-08-06 17:13:02

I'll go ahead and close this out then since it's not really a bug, but if you want you can place a feature request for "make generators work even if no received audio available" or something like that.