Re: [Freeswitch-dev] Unexpected call drop on PSTN network
Hi, The issue got resolved. I did not set "ignore_early_media=true" in the dial string. Everything is fine after that. Thanks! On Tue, Mar 26, 2013 at 5:04 AM, < freeswitch-dev-request@lists.freeswitch.org> wrote:
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Today's Topics:
1. Re: sofia_glue - sdp with m=image 0 (Anthony Minessale) 2. Unexpected call drop on PSTN network (Ashish gautam) 3. FreeSWITCH Weekly News and Notes (Michael Collins)
---------- Forwarded message ---------- From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-dev@lists.freeswitch.org Cc: Date: Fri, 22 Mar 2013 11:54:48 -0500 Subject: Re: [Freeswitch-dev] sofia_glue - sdp with m=image 0 The log would have helped explain that you are using proxy_media mode which is an important fact.
Please try latest HEAD. And from now on report issues like this to http://jira.freeswitch.org
Bugs should not be fixed via the mailing list.
On Fri, Mar 22, 2013 at 2:01 AM, Wolfgang Pichler <wpichler@yosd.at>wrote:
sorry - i do have already reconfigured the audiocodes gateway which was sending this - and it is already in production...
But i think the whole thing is also visible without trace, it is more a basic implementation question.
The question is - is it really a good idea to overwrite the audio port - when there is also a image port in the sdp ?
Here is the function i mean:
switch_status_t sofia_glue_tech_proxy_remote_addr(private_object_t *tech_pvt, const char *sdp_str) { const char *err; char rip[RA_PTR_LEN] = ""; char rp[RA_PTR_LEN] = ""; char rvp[RA_PTR_LEN] = ""; char *p, *ip_ptr = NULL, *port_ptr = NULL, *vid_port_ptr = NULL, *pe; int x; const char *val; switch_status_t status = SWITCH_STATUS_FALSE;
if (zstr(sdp_str)) { sdp_str = tech_pvt->remote_sdp_str; }
if (zstr(sdp_str)) { goto end; }
if ((p = (char *) switch_stristr("c=IN IP4 ", sdp_str)) || (p = (char *) switch_stristr("c=IN IP6 ", sdp_str))) { ip_ptr = p + 9; }
if ((p = (char *) switch_stristr("m=audio ", sdp_str))) { port_ptr = p + 8; }
if ((p = (char *) switch_stristr("m=image ", sdp_str))) { port_ptr = p + 8; }
if ((p = (char *) switch_stristr("m=video ", sdp_str))) { vid_port_ptr = p + 8; }
As you can see - it first checks for audio port - writes it to port_ptr, then checks for the image port - and if it does exists (also if it exists - but is zero - which is legal) - it does overwrite the audio port.
So calls with sdp
m=audio 12123 and m=image 0
will get dropped by freeswitch !
Thanks for you attention...
br, Wolfgang
2013/3/22 Anthony Minessale <anthony.minessale@gmail.com>
post a trace with "sofia global siptrace on" its too hard to diagnose without the exact trace.
On Thu, Mar 21, 2013 at 10:03 AM, Wolfgang Pichler <wpichler@yosd.at>wrote:
Hi all,
i have encountered a strange behaviour - and i think i have found a possible problem within sofia_glue
Following scenario:
Carrier sends INVITE with T38 Offer -> Freeswitch does forward offer -> asterisk does forward offer -> ATA
In this scenario it happend to me - that the T38 Offer (with corrent m=image and port in sdp) got forwarded by freeswitch to asterisk, asterisk did forwarded without the T38 Offer (don't know why) - so also without the m=image sdp part - the ATA did answered correctly (without m=image part) -> asterisk did created a new sdp with m=image 0, without T38 parts - and returned this to freeswitch.
Freeswitch did hangup the call - because sofia_glue did extracted the port 0 as audio port - and this is not a legal port...
Sending port 0 is according to rfc legal - so freeswitch should ignore it in this case.
the question is - if both values are given - which one is the value we need ? I think the audio port is the port which is of higher priority.
So the workflow should be: - Try to extract audio port from m=audio part. - If you got it - and it is a valid port then go on - else try to extract audio port from m=image part...
The function in question is sofia_glue_tech_proxy_remote_addr - in sofia_glue.c
Is it generaly a good idea to use the m=image part as audio port ? Why is there no direct image port member in the pvt structure as with the video port ?
Hope someone can help here...
br, Wolfgang
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
---------- Forwarded message ---------- From: Ashish gautam <ashish@nms.co.in> To: freeswitch-dev@lists.freeswitch.org Cc: Date: Fri, 22 Mar 2013 16:23:43 +0530 Subject: [Freeswitch-dev] Unexpected call drop on PSTN network Hi,
I am facing a strange issue with making outgoing calls to PSTN netwrok through freetdm. The calls are successfully made to some Telecom networks while they just drop abruptly and hangup on other networks. The calls are being generated through inbound Even Socket originate action and put in a public context which executes a perl script in the dialplan.
Please help me out. Thanks in advance.
-- Ashish
---------- Forwarded message ---------- From: Michael Collins <marketing@cluecon.com> To: freeswitch-users@lists.freeswitch.org, freeswitch-dev@lists.freeswitch.org, freeswitch-cluecon@lists.freeswitch.org Cc: Date: Mon, 25 Mar 2013 16:34:03 -0700 Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes Greetings!
We are back after a brief hiatus. Our friend and colleague Brian West took some much-deserved time off last week so the rest of us were a bit busier than usual. Brian does an amazing amount of work on FreeSWITCH, CudaTel, and ClueCon, so we're definitely glad he's back to work!
On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_20>we talked about Debian packaging with a little VoIP security and ZRTP thrown in for good measure. Interestingly, a number of people stayed on the call until 6pm EST! You are welcome to stay in the public conference room for as long as you like. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2013_03_27>we have Mark Crane scheduled to give us an update on FusionPBX. It's been about 12 months since we last heard from him. We are looking forward to seeing the improvements that have been added in the past year.
We are also gearing up for ClueCon 2013 <http://www.cluecon.com> so save the date: August 6-8, 2013. If you have any questions about being a speaker, sponsor, or attendee then by all means contact us at this email address. Remember that WebRTC is a big topic in the news right now and we are looking for folks to talk about WebRTC in their open source telephony projects and solutions.
Have a great week and we'll talk to you again in April!
-- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE
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Ashish gautam