[Freeswitch-dev] Unexpected call drop on PSTN network
Ashish gautam
ashish at nms.co.in
Tue Mar 26 12:12:04 MSK 2013
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 at lists.freeswitch.org> wrote:
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> 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 at gmail.com>
> To: freeswitch-dev at 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 at 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 at 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 at 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
>>>>
>>>>
>>>> _________________________________________________________________________
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>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
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>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
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>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
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>>>
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>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
>>> Official FreeSWITCH Sites
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>>> http://www.cluecon.com
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>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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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 at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> ---------- Forwarded message ----------
> From: Ashish gautam <ashish at nms.co.in>
> To: freeswitch-dev at 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 at cluecon.com>
> To: freeswitch-users at lists.freeswitch.org,
> freeswitch-dev at lists.freeswitch.org,
> freeswitch-cluecon at 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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