should work.
Any day I learn something about SIP is a good day. :-)
Thanks.
Jim
- Original Message ----
From: Michael Jerris
To: Jim Thomas ; sofia-sip-devel
Sent: Thu, May 20, 2010 8:35:09 AM
Subject: Re: [Sofia-sip-devel] Header created using SIPTAG_HEADER_STR appears
after Content-Length
Hello,
I am experimenting with adding support for SIP 302 to redirect an inbound call
when the PBX is configured to forward the call.
The ITSP requires that the Contact header include the number where the call
should be redirected, and a Diversion header must be added that identifies the
valid
Hello,
I am using Sofia-SIP 1.12.10 and it works great. Thank you
for your open source contribution.
I use an ITSP where, until recently, a DNS lookup by Sofia-SIP
would obtain a single SRV record. Outbound SIP calls work as
expected every time.
The ITSP now wants to distribute the SIP traffic
was included in the local SDP string
passed to SOATAG_USER_SDP_STR(). The result is that the ITSP continues sending
RFC 2833 RTP Events after Session Refresh and does not revert to in-band DTMF
audio.
- Original Message ----
From: Jim Thomas
To: sofia-sip-devel@lists.sourceforge.net
Sent:
Hello,
Using Sofia-SIP 1.12.10, I send and INVITE that includes 'telephone-event' to
enable RFC 2833 RTP Events. Tcpdump confirms it is present on the wire:
v=0
o=- 8101859643205583490 1396692759725295989 IN IP4 192.168.0.21
s=-
c=IN IP4 192.168.0.24
t=0 0
m=audio 30062 RTP/AVP 18 1
ing to identify where this is occurring and reconfigure the
SBC, which may not always be possible.
Thanks.
Jim
- Original Message
From: Jerry Richards
To: sofia-sip-devel@lists.sourceforge.net
Cc: Jim Thomas ; Inca Rose
Sent: Friday, July 31, 2009 2:23:54 PM
Subject: RE: REGISTER Ref
Hello Jerry,
It looks like you and I have run into the same issue. Please see my recent
post 'Registration issue with SIP server behind SBC'.
I think Sofia-SIP is doing the right thing. From RFC 3261:
The 200 (OK) response from the registrar contains a list of Contact
fields enumerating a
Hello,
Pekka Pessi wrote on 22 Feb 2006:
> Nua goes through the list of contacts in response to REGISTER,
> if it founds a URI that matches with a URI in original REGISTER
> request it calculates the expiration time of contact with
> sip_contact_expires().
I am using Sofia-SIP v1.12.10
For an inbound call, when I call nua_bye(), the Sofia-SIP stack inserts an IP
address in the Request-URI for the BYE.
How can I control the nua_bye() Request-URI field so I can force it to use a
desired URI instead of an IP address?
Thanks.
__
I need to add code to make my Sofia-SIP integration more resilient if periodic
re-registration fails.
Please help me think about how best to approach this.
Thanks.
How to recover from failed auto re-registration due to DNS error
Jim Thomas
Fri, 27 Mar 2009 11:15:05 -0700
Hello,
When
Hello,
When successfully registered with the ITSP using Sofia-SIP v1..12.10, I
observe reasonable periodic re-registration events in my application log:
12:25:52 INF: auto re-registration: 200 OK
12:26:56 INF: auto re-registration: 200 OK
12:28:12 INF: auto re-registration: 100 Request Au
Pekka,
I successfully built and ran your release candidate sofia-sip-1..12.9pre10rc2
on my ARM9 (Linux, gcc) platform this morning.
My application is not very far along yet, so this is not a thorough test, but I
wanted to pass along that so far this release candidate looks good as an
upgrade f
Jen wrote:
> I could try grabbing the latest from Darcs, but I'm nervous
> about using that, as it's not a "stable release", and I really
> do need stability.
I second the motion for a fresh stable release.
Jim
- Original Message
From: Michael Jerris <[EMAIL PROTECTED]>
To: Je
Hello,
Sofia-SIP v1.12.9 was released on 5 Jun 2008.
When is the next stable release anticipated?
Thanks.
Jim
-
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Hello,
I am experimenting with Sofia-SIP and like it very much.
I am currently carrying a 'call' structure around via the nua_hmagic_t hmagic
pointer. It works well.
However, it looks like this same hmagic pointer may also be used in some
non-call situations such as registration.
In anticipation
Hello,
I am trying to make the case for using Sofia SIP.
I am aware that Sofia SIP is used in the Nokia Internet Tablet, and is also
used by FreeSWITCH.
Is Sofia SIP used in other products? Can anyone share any success stories?
Is Sofia SIP maintained exclusively (or nearly exclusively) by Nokia,
(Sorry if this is a repeat, I got a bounce notification on the first attempt.)
Hello,
Is the Sofia-SIP stack run through any SIP torture tests prior to release?
What kind of stress and compatibility testing is typically done?
Thanks.
Jim
-
either using Sofia-SIP or has plans
to integrate it in the future.
Do you regard Sofia-SIP as suitable for this purpose? Any and all
opinions/guidance is most welcome.
Jim
- Original Message
From: Pekka Pessi <[EMAIL PROTECTED]>
To: Jim Thomas <[EMAIL PROTECTED]>
Cc: sof
Hello,
I am considering using Sofia-SIP in a commercial product.
Open source projects tend to come and go.
Is Sofia-SIP likely to be actively maintained and further developed into the
future?
What is the ongoing level of commitment to Sofia-SIP by Nokia and others?
Thanks.
Jim
-
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