Hello All,
I am having a bit peculiar problem with Asterisk 11 for a carrier. This carrier shares quite some information in SDP header, which should not be the problem, however what happen is as follow:


Carrier—-> (INVITE) -> *SIP Proxy -> Asterisk 11 -> Answer()* -> right after answering call drops… Carrier send a BYE with (cause 79: service or option not implemented).

*NOTE: Please refer to complete SIP traces attached. *
*
*
*Also Note:*
*Carrier*: 62.61.147.214
*Proxy*: 77.X.X.X:5060
*Asterisk11*: 77.X.X.X:5080

*Here is Invite SDP from Carrier -> Proxy -> Asterisk 11*

INVITE sip:69609000@77.X.X.X SIP/2.0
v=0
o=AudiocodesGW 1638819008 1638818710 IN IP4 62.61.147.214
s=Phone-Call c=IN IP4 77.X.X.X
t=0 0
m=audio 53372 RTP/AVP 8 118 18
a=rtpmap:8 PCMA/8000
a=rtpmap:118 PCMA/8000
a=gpmd:118 vbd=yes a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no a=ptime:20
a=sendrecv a=rtcp:53373 IN IP4 77.X.X.X
m=image 56854 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

*SDP:After Answered by Asterisk 11*
v=0
o=root 164966782 164966782 IN IP4 77.X.X.X
s=Asterisk v11.0.1
c=IN IP4 77.X.X.X
t=0 0
m=audio 12636 RTP/AVP 18 8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
*m=image 0 udptl t38*

I have tired by disabling/unloading fax modules as *I am not using* them but no results. Secondly, also tried tweaking of udptl ever-odd nothing worked.

The same carrier works for Asterisk 1.6.X and the only difference I have notice so far is the above underlined line in Answered SDP -> m=image 0
udptl t38. I think if I some how do not advertise udptl here i would be able to avoid this scenario. I have tried multiple ways to strip off SDP
from incoming INVITE at SIP proxy level but it is not SDP wise enough.


*Note:*

In Asterisk 1.6 => WARNING[32671]: chan_sip.c:8833 process_sdp:
Unsupported SDP media type in offer: image 59978 udptl t38
In Asterisk 11 => WARNING[18748][C-0000002f]: chan_sip.c:10277 process_sdp:
Failed to initialize UDPTL, declining image stream

Given the limitations around Asterisk‘s TLS support, and all the benefits of using a SIP proxy, I’ve put together a rough guide about how to use the repro SIP proxy as a front-end for Asterisk connectivity with TLS peers:

http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch

It works for TLS from phones, but also for full federated SIP with any other SIP-enabled domain on the public Internet.

* repro does all the connectivity work (certificate validation, etc) and registration service

* Asterisk sits in the background and provides applications (voicemail, queues, etc)

Any feedback, questions or discussion about this is very welcome.

Hi everybody, I want to use asterisk for build a SIP Proxy and REGISTAR, Is
AsteriskNow a Sip Proxy already? or, do i need to build it?

How does asterisk solves the issues of a SIP USER AGENT Behind a NAT?

Does asterisk implements RFC 5626?

Does asterisk implements STUN, TURN and ICE?

Thanks for your time.

Sorry for my english.

Hi All;

I need to use Asterisk for 20 000 users, so which asterisk version to be used? Is there asterisk version that supports 20,000 users on one hardware machine?

Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to handle 20 000 users, and concurrent calls 2000? Or I need multiple servers, how much?

If I am going to use multiple servers (until now I do not know how much, and I do not know if the barrier will be the asterisk software or the hardware), then do I have to use special SIP proxy or I have to use load balancer)? In this case, I have to use asterisk Database (so all the servers will read/write from the database)?

What about AsteriskNow, can it support?

Regards
Bilal

On 05/10/2012 09:39 AM, Arif Hossain wrote:
> I have following sip account :
>
> Name/username Host Dyn
> Forcerport ACL Port Status Description
> demo-alice/demo-alice 192.168.7.47 D
> N 1080 Unmonitored
> demo-bob/demo-bob 192.168.7.47 D
> N 5060 Unmonitored
>
> and i have set up the following extensions for them:
>
> ASTERISK_IP=192.168.7.39
>
> [users]
> exten=>6001,1,Dial(SIP/demo-alice,20)
> exten=>6002,1,Dial(SIP/demo-bob,20)
>
> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
> exten => _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
> exten => _.,n,HangUp()u
>
> [macro-uri-dial]
> exten=>s,n,NoOp(Calling as SIP address: ${ARG1})
> exten=>s,n,Dial(SIP/${ARG1},60)
>
>
> But if i dial sip uri the call does not happen. asterisk cli shows
> extension is rejected.

Asterisk is not a SIP proxy. If you are entering a SIP URI into your
phone, and that URI does not resolve to the Asterisk server as its
target, then the INVITE request sent by the phone should not even be
sent to Asterisk at all (it should go to wherever the URI resolves to).