In the past week, I have merged patches into Asterisk trunk that provide new features for chan_sip. These features will be available in Asterisk 1.6.
TCP and TLS support
In the past, Asterisk has only had support for UDP as a transport for SIP signaling. Asterisk 1.6 will have support for both TCP and TLS, as well.
This work was done by Brett Bryant and James Golovich, who were both funded by Digium for their work.
SIP Session Timers (RFC 4028)
Asterisk 1.6 will also have support for SIP Session Timers, as defined by RFC 4028. These changes prevent stuck SIP sessions that were not properly torn down due to network or endpoint failures during an established SIP session.
This work was done by Raj Jain. It was funded by John Todd from TalkPlus, Inc., as well as JR Richardson of Ntegrated Solutions.
Both of these features are major steps forward for SIP support in Asterisk!
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I have a query regarding TCP support for Asterisk 1.6. Does Asterisk 1.6 support TCP when it acts as a client ?
Yes, it does.
what about OCS 2007 compatibility?
What I really would like to see is integration of zfone into asterisk, and not as some third party patch, but by default.
We need proper call security, particularly when going over semi-public networks, like hotel WiFi LANs, etc.
and more general information about the project:
Can Asterisk 1.6 saemlessly integrate with Microsoft OCS?
Hi, I have install new version of Asterisk 1.6.x I can not see any screen from which I can ADD SIPP extension. How shall I add a sip extension from web based application ? It could be easily done in 1.4.x version.
Which kind of Microsoft OCS integration do you like to have?
Will it support e.164 number processing by OCS? Will SIP endpoints register with OCS?
What OCS support does it have?
OCS is rubbish. Shame it supports “SIP” in their way.
Willit support full SRV (multiple records)? Not just the first record?
Cisco, Avaya, Nortel, etc. all support SIP in their “own way”. SIP has so many ways of doing any one thing… let alone a set of things. Given this, interoperability is *really* hard — and if devices from different vendors don’t interoperate — blame the IETF, not the vendors.
I am very new to SIP and I heard that asterisk supports SIP.How does astersk supports SIP?
Do we need to plug in a sip proxy to asterisk to make voip calls.
Does 1.6 support Wi-fi MultiMedia, (WMM), for Polycom Spectraling 8002 wireless phones?
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Regarding SIPP tls connection with Asterisk
Postby shivnamha » Thu Dec 08, 2011 7:26 am
I configured all the setting on Asterisk for activating tls connection . Then i tried to access asterisk server using SIPp command with tls option. then m getting error .
here is the command
sipp -sn uac -d 2 -s 1000 10.11.20.31:5061 -l 1 -m 1 -max_retrans 1 -t l1 -tls_cert /root/new_certificate.pem -tls_key /root/new_key.pem
Error is Unable to connect a TCP socket.
Use ‘sipp -h’ for details, errno = 111 (Connection refused).
please Help regarding this issue . But i created a service of tls on 5061 port but that service is not listening on Asterisk.
thanks in Adavance.