[Freeswitch-users] Question about rtp-timeout-sec variable
Hello, I have 2 instances of FS: one controlled by my application (making calls with TCP commands, recording sessions, listening to events etc) and one acting as a remote gateway to which all users register. When I leave the default values of rtp-timeout-sec and brutally kill x-lite during conversation, the 'hangup' event with 'media_timeout' cause is obviously sent after the default 5 minutes (and until then, the other leg is still connected to a 'dead' channel). The question is: which FS instance is responsible for terminating the connection after timeout? Only the 'remote' FS instance config seems to work. I thought that the shortest configured value should cause the timeout, but it's not the case. Am I missing something, or is this the correct behavior? Regards, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Question-about-rtp-timeout-sec-variable-tp4050650p4050650.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Users hanged up for unknown reason
Hi, I have a strange problem. I control FS with commands sent by tcp in response to events published via tcp. I do something like: 1) call 1st user 2) call 2nd user 3) 1st and 2nd talk 4) call another user 5) 1st and another talk etc... Sometimes (quite regularly) users are hanged up (with cause NORMAL_CLEARING) even if they do not hangup manually. I pasted one such scenario in pastebin (http://pastebin.freeswitch.org/10955), it includes logs from commands sent by me and events received from FS. Could someone take a look and see what am I doing wrong? The scenario includes 3 users 1st user (Unique-ID: f076261a-4537-40f2-b46d-933141320314) is supposed to be connected all the time but gets diconnected 2nd user (Unique-ID: ebdfb398-ec82-4760-9f79-81364e0f37b6) is supposed to talk for a few seconds and get killed 3rd user (Unique-ID: d5cd839e-793c-4b3c-adda-327841672a5f) is supposed to work like 2nd user All of them are simulated by dialplan extensions (using answer and playback tools), but the same thing happends for xlite or cisco phone. Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Users-hanged-up-for-unknown-reason-tp3937601p3937601.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] "Can not record session. Media not enabled on channel."
Hi, Unfortunately getting the newest version did not solve the problem: "Can not record session. Media not enabled on channel." error still appears sometimes. MA Maciej Aniserowicz wrote: > > Correct - compiled but did not run. Works fine now. > > I'll see if the error shows up again and let you know if it does. > Thanks, > MA > > > > Anthony Minessale wrote: >> >> won't compile or won't run? >> maybe you should try rebuilding it. >> >> >> On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz < >> maciej.aniserow...@gmail.com> wrote: >> >>> Sorry, trunk does not compile on win7, here are the details: >>> >>> >>> rev.15247 >>> >>> --- >>> Microsoft Visual C++ Debug Library >>> --- >>> Debug Assertion Failed! >>> >>> >>> >>> >>> >>> - Original Message - >>> *From:* [hidden >>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=0> >>> *To:* [hidden >>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=1> >>> *Sent:* Monday, October 26, 2009 10:32 PM >>> *Subject:* Re: [Freeswitch-users] "Can not record session. Media not >>> enabled on channel." >>> >>> >>> >>> On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden >>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=0> >>> > wrote: >>> >>>> >>>> Yes, I can confirm - this exact error occurs each time when I start >>>> recording >>>> before the call is answered (just after sending ORIGINATE command) - >>>> but I >>>> think that's completely understandable that media is not ready for an >>>> unanswered call. >>>> But... is there any other event that guarantees media to be ready? >>>> >>>> Update to latest SVN and try again. >>> -MC >>> >>> >>> ___ >>> FreeSWITCH-users mailing list >>> [hidden >>> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=1> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> -- >>> View this message in context: Re: [Freeswitch-users] "Can not record >>> session. Media not enabled on >>> channel."<http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html> >>> >>> Sent from the freeswitch-users mailing list >>> archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at >>> Nabble.com. >>> >>> ___ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> 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_miness...@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:8...@conference.freeswitch.org >> iax:gu...@conference.freeswitch.org/888 >> googletalk:conf+...@conference.freeswitch.org >> pstn:213-799-1400 >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3936705.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] "Can not record session. Media not enabled on channel."
Correct - compiled but did not run. Works fine now. I'll see if the error shows up again and let you know if it does. Thanks, MA Anthony Minessale wrote: > > won't compile or won't run? > maybe you should try rebuilding it. > > > On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz < > maciej.aniserow...@gmail.com> wrote: > >> Sorry, trunk does not compile on win7, here are the details: >> >> >> rev.15247 >> >> --- >> Microsoft Visual C++ Debug Library >> --- >> Debug Assertion Failed! >> >> >> >> >> >> - Original Message - >> *From:* [hidden >> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=0> >> *To:* [hidden >> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3899478&i=1> >> *Sent:* Monday, October 26, 2009 10:32 PM >> *Subject:* Re: [Freeswitch-users] "Can not record session. Media not >> enabled on channel." >> >> >> >> On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden >> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=0> >> > wrote: >> >>> >>> Yes, I can confirm - this exact error occurs each time when I start >>> recording >>> before the call is answered (just after sending ORIGINATE command) - but >>> I >>> think that's completely understandable that media is not ready for an >>> unanswered call. >>> But... is there any other event that guarantees media to be ready? >>> >>> Update to latest SVN and try again. >> -MC >> >> >> ___ >> FreeSWITCH-users mailing list >> [hidden >> email]<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3895104&i=1> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> View this message in context: Re: [Freeswitch-users] "Can not record >> session. Media not enabled on >> channel."<http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html> >> >> Sent from the freeswitch-users mailing list >> archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at >> Nabble.com. >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org > pstn:213-799-1400 > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3906568.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] "Can not record session. Media not enabled on channel."
Sorry, trunk does not compile on win7, here are the details:
rev.15247
---
Microsoft Visual C++ Debug Library
---
Debug Assertion Failed!
Program: ...ev\Projects\External\FreeSWITCH\Original\Debug\FreeSwitch.exe
File: f:\dd\vctools\crt_bld\self_x86\crt\src\dbgheap.c
Line: 1317
Expression: _CrtIsValidHeapPointer(pUserData)
For information on how your program can cause an assertion
failure, see the Visual C++ documentation on asserts.
(Press Retry to debug the application)
---
Abort Retry Ignore
---
VS Call stack:
ntdll.dll!77ccfadc()
[Frames below may be incorrect and/or missing, no symbols loaded for
ntdll.dll]
ntdll.dll!77c9272c()
ntdll.dll!77c5e1ef()
msvcr90d.dll!_free_dbg_nolock(void * pUserData=0x00664b88, int nBlockUse=1)
Line 1317 + 0x9 bytes C++
msvcr90d.dll!_free_dbg(void * pUserData=0x00664b88, int nBlockUse=1) Line
1258 + 0xd bytes C++
msvcr90d.dll!free(void * pUserData=0x00664b88) Line 49 + 0xb bytes C++
> FreeSwitch.dll!switch_core_mime_add_type(const char * type=0x003bcd1c, const
> char * ext=0x003bcd37) Line 748 + 0xc bytes C
FreeSwitch.dll!load_mime_types() Line 791 C
FreeSwitch.dll!switch_core_init(unsigned int flags=65, switch_bool_t
console=SWITCH_TRUE, const char * * err=0x003bf68c) Line 1244 C
FreeSwitch.dll!switch_core_init_and_modload(unsigned int flags=65,
switch_bool_t console=SWITCH_TRUE, const char * * err=0x003bf68c) Line 1454 +
0x11 bytes C
FreeSwitch.exe!main(int argc=1, char * * argv=0x02144c40) Line 764 + 0x23
bytes C
FreeSwitch.exe!__tmainCRTStartup() Line 586 + 0x19 bytes C
FreeSwitch.exe!mainCRTStartup() Line 403 C
kernel32.dll!77713677()
ntdll.dll!77c39d72()
ntdll.dll!77c39d45()
Error occurs in :
SWITCH_DECLARE(switch_status_t) switch_core_mime_add_type(const char *type,
const char *ext)
{
const char *check;
switch_status_t status = SWITCH_STATUS_FALSE;
switch_assert(type);
switch_assert(ext);
check = (const char *) switch_core_hash_find(runtime.mime_types, ext);
if (!check) {
char *ptype = switch_core_permanent_strdup(type);
char *ext_list = strdup(ext);
int argc = 0;
char *argv[20] = { 0 };
int x;
switch_assert(ext_list);
if ((argc = switch_separate_string(ext_list, ' ', argv, (sizeof(argv) /
sizeof(argv[0]) {
for (x = 0; x < argc; x++) {
if (argv[x] && ptype) {
switch_core_hash_insert(runtime.mime_types, argv[x], ptype);
}
}
status = SWITCH_STATUS_SUCCESS;
}
free(ext_list); // <--- HERE
}
return status;
}
- Original Message -
From: mercutioviz [via freeswitch-users]
To: Maciej Aniserowicz
Sent: Monday, October 26, 2009 10:32 PM
Subject: Re: [Freeswitch-users] "Can not record session. Media not enabled on
channel."
On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden email]> wrote:
Yes, I can confirm - this exact error occurs each time when I start
recording
before the call is answered (just after sending ORIGINATE command) - but I
think that's completely understandable that media is not ready for an
unanswered call.
But... is there any other event that guarantees media to be ready?
Update to latest SVN and try again.
-MC
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Re: [Freeswitch-users] "Can not record session. Media not enabled on channel."
Yes, I can confirm - this exact error occurs each time when I start recording before the call is answered (just after sending ORIGINATE command) - but I think that's completely understandable that media is not ready for an unanswered call. But... is there any other event that guarantees media to be ready? mercutioviz wrote: > > On Fri, Oct 23, 2009 at 12:36 AM, Maciej Aniserowicz < > maciej.aniserow...@gmail.com> wrote: > >> >> The dialplan is very simple: >> >> >>> expression="^11\d*$"> >> >> >> >> >>> data="local_stream://my_music"/> >> >> >> >> Before debugging I have another question: I start recording in event >> handler >> for ChannelAnswer event. Is it possible that it's too soon to start >> recording? Maybe I should start recording in some other event? >> > > That would be an odd scenario but maybe. It would be best if you could > catch > it in the act so that we could see exactly what is happening. The other > thing you could do is deliberately start recording on the channel prior to > answering and see if you always get the error. In other words, try to make > it fail under a certain set of circumstances to see if your theory is > correct. > -MC > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3890610.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] "Can not record session. Media not enabled on channel."
The dialplan is very simple: Before debugging I have another question: I start recording in event handler for ChannelAnswer event. Is it possible that it's too soon to start recording? Maybe I should start recording in some other event? MAniserowicz mercutioviz wrote: > > What's in the dialplan for this channel? Is bypass-media or proxy-media > set > to true? Do a debug trace and post it in pastebin. > -MC > > On Tue, Oct 20, 2009 at 3:30 AM, Maciej Aniserowicz < > maciej.aniserow...@gmail.com> wrote: > >> >> Hello, >> I am using the same set of extensions for testing the system during >> development, they include XLite, Cisco sip phone and several extensions >> that >> just play some audio file. >> Sometimes, very rarely, this message "Can not record session. Media not >> enabled on channel." appears on FS console. >> Like I wrote before, I don't change any codec settings and always use the >> same set of devices/emulators. >> What can cause this message? >> >> Thanks, >> Maciej Aniserowicz >> > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3877285.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] "Can not record session. Media not enabled on channel."
Hello, I am using the same set of extensions for testing the system during development, they include XLite, Cisco sip phone and several extensions that just play some audio file. Sometimes, very rarely, this message "Can not record session. Media not enabled on channel." appears on FS console. Like I wrote before, I don't change any codec settings and always use the same set of devices/emulators. What can cause this message? Thanks, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3857858.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bad sound quality while eavesdropping
Nope, I wanted to make sure that this is indeed a bug. I opened an issue in JIRA before regarding some other matter and it turned out to be my mistake, so I decided to try mailing list first this time. MA Brian West wrote: > > Did you open a jira and attach all the info? > > /b > > On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote: > >> Yes, I confirmed that with Wireshark (filter "rtp and ip.src == >> ). RTP packets are sent every 20ms. >> >> MAniserowicz >> > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3808860.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Bad sound quality while eavesdropping
Yes, I confirmed that with Wireshark (filter "rtp and ip.src == ). RTP packets are sent every 20ms. MAniserowicz - Original Message - From: Michael Jerris (via Nabble) To: Maciej Aniserowicz Sent: Monday, October 12, 2009 12:21 AM Subject: Re: [Freeswitch-users] Bad sound quality while eavesdropping can you confirm from an rtp packet trace that they are all really sending 20ms? Mike On Oct 10, 2009, at 6:04 AM, Maciej Aniserowicz wrote: > > Hi, > Here are the messages with a:ptime parameter. All the calls are > started by > commands sent through socket. > I'm not sure if this is all information you need, please let me know > if > something is missing here and I'll post that. > > 1) starting connection with x-lite (number 2003, the eavesdropper): > > INVITE sip:[hidden email]:60188;rinstance=80b8f8d92af87cd2 SIP/ > 2.0 > Via: SIP/2.0/UDP > 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K > Max-Forwards: 69 > From: "MyApp" ;tag=jpQ6D7D2jUXvF > To: > Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff > CSeq: 121465610 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 447 > Remote-Party-ID: "MyApp" > ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 2223565947735016740 3096553520713245589 IN IP4 > 192.168.3.159 > s=FreeSWITCH > c=IN IP4 192.168.3.159 > t=0 0 > m=audio 29966 RTP/AVP 0 115 107 9 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:115 G7221/32000 > a=fmtp:115 bitrate=48000 > a=rtpmap:107 G7221/16000 > a=fmtp:107 bitrate=32000 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > > 2) starting connection with cisco ip phone (number 2006, first leg of > eavesdropped session): > > INVITE sip:[hidden email]:5060;user=phone SIP/2.0 > Via: SIP/2.0/UDP > 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p > Max-Forwards: 69 > From: "MyApp" ;tag=Q3N2pe2K47ctS > To: > Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff > CSeq: 121465616 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 447 > Remote-Party-ID: "MyApp" > ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 8453497903781974949 2294419114567885490 IN IP4 > 192.168.3.159 > s=FreeSWITCH > c=IN IP4 192.168.3.159 > t=0 0 > m=audio 17670 RTP/AVP 0 115 107 9 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:115 G7221/32000 > a=fmtp:115 bitrate=48000 > a=rtpmap:107 G7221/16000 > a=fmtp:107 bitrate=32000 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > > 3) starting connection with extension playing a file (number , > second > leg of eavesdropped session): > > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS > From: "FreeSWITCH" > ;tag=091j2Q0Fre8vp > To: ;tag=U7t5Xt51rB64Q > Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8 > CSeq: 121465623 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk,
Re: [Freeswitch-users] Bad sound quality while eavesdropping
Hi, Here are the messages with a:ptime parameter. All the calls are started by commands sent through socket. I'm not sure if this is all information you need, please let me know if something is missing here and I'll post that. 1) starting connection with x-lite (number 2003, the eavesdropper): INVITE sip:2...@192.168.3.100:60188;rinstance=80b8f8d92af87cd2 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K Max-Forwards: 69 From: "MyApp" ;tag=jpQ6D7D2jUXvF To: Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff CSeq: 121465610 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 447 Remote-Party-ID: "MyApp" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 2223565947735016740 3096553520713245589 IN IP4 192.168.3.159 s=FreeSWITCH c=IN IP4 192.168.3.159 t=0 0 m=audio 29966 RTP/AVP 0 115 107 9 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 2) starting connection with cisco ip phone (number 2006, first leg of eavesdropped session): INVITE sip:2...@192.168.2.106:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p Max-Forwards: 69 From: "MyApp" ;tag=Q3N2pe2K47ctS To: Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff CSeq: 121465616 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 447 Remote-Party-ID: "MyApp" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 8453497903781974949 2294419114567885490 IN IP4 192.168.3.159 s=FreeSWITCH c=IN IP4 192.168.3.159 t=0 0 m=audio 17670 RTP/AVP 0 115 107 9 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:107 G7221/16000 a=fmtp:107 bitrate=32000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 3) starting connection with extension playing a file (number , second leg of eavesdropped session): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS From: "FreeSWITCH" ;tag=091j2Q0Fre8vp To: ;tag=U7t5Xt51rB64Q Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8 CSeq: 121465623 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 263 v=0 o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4 192.168.3.159 s=FreeSWITCH c=IN IP4 192.168.3.159 t=0 0 m=audio 30086 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Anthony Minessale wrote: > > you probably have some device lying about ptime everywhere > look at a sip trace an pay especially close attention to ptime:x param in > sdp > if you don't understand this just attach it here > > execute the following at the cli > sofia profile internal siptrace on > sofila loglevel debug > > > > On Thu, Oct 8, 2009 at 8:27 AM, Maciej Aniserowicz < > maciej.aniserow...@gmail.com> wrote: > >> >> It's the same on the trunk (the last rev I used was not so old anyway). >> >> Codecs are the same on both legs: >> read codec/read rate: PCMU 8000 >> write codec/write rate: PCMU8000 >> >> MA >> >> >> >> >> Michael Jerris wrote: >> > >> > What codecs are all the cal
Re: [Freeswitch-users] gateway FS informs it's client FS about users hanged up with a long delay
Hello,
The issue is resolved. I feel stupid, because Michael Jerris was right the
first time. Setting external_rtp_ip and external_sip_ip to $${local_ip_v4} made
it work.
But the strange thing is: it SOMETIMES worked before without any delay, which
'should not be possible', because the original IP was my external ip and the
BYE message was sent straight to it. And there is no way it could reach the
target 'internal' FS, because it runs on virtual machine, and no ports are
forwarded on my router.
Any thoughts? Why this could (rarely) work even with the previous config?
Thanks to both of you for your answers.
MA
- Original Message -
From: mercutioviz (via Nabble)
To: Maciej Aniserowicz
Sent: Thursday, October 08, 2009 7:06 PM
Subject: Re: [Freeswitch-users] gateway FS informs it's client FS about users
hanged up with a long delay
On Thu, Oct 8, 2009 at 6:18 AM, Maciej Aniserowicz <[hidden email]> wrote:
Both of the instances are run on the same machine, i just changed the
default
ports they use. Can anything else cause this strange behavior?
MA
Did a packet capture yield any clues? That is, were you able to confirm that
each instance sent and received all the packets that you believe they should
have sent and received? The reason I ask is so that you don't end up chasing a
ghost because you made an assumption somewhere in your troubleshooting.
-MC
Â
Michael Jerris wrote:
>
> Incorrect NAT configuration so one of the boxes is not actually
> getting a BYE.
>
>
> On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
>
>> Hi,
>> When I use two FreeSWITCH instances ('internal' and 'external'), all
>> users register to the 'external' instance which acts as a gateway by
>> 'internal' instance (which in turn is controlled by my applicaiton
>> with commands sent by socket).
>> When user hangs up, the 'hanged up' event is propagated to the
>> 'internal' instance after a long time (~3 minutes) instead of being
>> propagated immediately.
>> What can cause this issue?
>
>
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Re: [Freeswitch-users] Bad sound quality while eavesdropping
It's the same on the trunk (the last rev I used was not so old anyway). Codecs are the same on both legs: read codec/read rate: PCMU 8000 write codec/write rate: PCMU8000 MA Michael Jerris wrote: > > What codecs are all the call legs using, also, please try current svn > trunk. > > Mike > > On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote: > >> >> Sorry about posting several questions at once, I wasn't aware it's >> "rude". >> Let's concentrate on this issue then. >> >> I use FS rev 14994. Phones on extensions: >> 1) x-lite >> 2) cisco sip phone >> 3) audio played by fs to the extension being eavesdropped >> >> I did not change any codec configuration, I just use the standard >> one that >> comes with both FS and the phones. >> Some time ago someone on FS irc channel told me that this is just >> how FS >> eavesdropping works... from your response I understand that this is >> not >> entirely true? >> >> Maciej Aniserowicz >> >> >> >> Anthony Minessale wrote: >>> >>> That's is a somewhat vague position. >>> >>> You did not mention which version of FreeSWITCH you are running, the >>> phones >>> being used in your example, your configuration, the codecs in use >>> etc. >>> >>> BTW, >>> I think you should only ask one question at a time on this list. >>> The list >>> is run by volunteers and it's sort of rude to expect 3 or 4 threads >>> to be >>> tended to concerning the same one individual. >>> >>> >>> 2009/10/5 Maciej Aniserowicz >>> >>>> Hello, >>>> When I use eavesdropping in FreeSWITCH, the sound quality is >>>> really bad. >>>> Is >>>> there any way to improve it? Is this a known problem? >>>> Br/ >>>> Maciej Aniserowicz >>>> > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Recording creates a 388-byte long file and deletes it
Yes, I know that FS deletes short files. I just don't know why the file is so small... it is always 388 bytes, no matter how long the session lasts. MA Michael Jerris wrote: > > switch_ivr_async.c:480 > > On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote: > >> Hi, >> When I record a call in FS, it only creates a 388-byte-long wav >> file. The conversation is no written there, and FS deletes the file >> when the session finishes. >> What can cause this strange behavior? > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Recording-creates-a-388-byte-long-file-and-deletes-it-tp3768541p3787968.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] gateway FS informs it's client FS about users hanged up with a long delay
Both of the instances are run on the same machine, i just changed the default
ports they use. Can anything else cause this strange behavior?
MA
Michael Jerris wrote:
>
> Incorrect NAT configuration so one of the boxes is not actually
> getting a BYE.
>
>
> On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
>
>> Hi,
>> When I use two FreeSWITCH instances ('internal' and 'external'), all
>> users register to the 'external' instance which acts as a gateway by
>> 'internal' instance (which in turn is controlled by my applicaiton
>> with commands sent by socket).
>> When user hangs up, the 'hanged up' event is propagated to the
>> 'internal' instance after a long time (~3 minutes) instead of being
>> propagated immediately.
>> What can cause this issue?
>
>
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Re: [Freeswitch-users] Bad sound quality while eavesdropping
Sorry about posting several questions at once, I wasn't aware it's "rude". Let's concentrate on this issue then. I use FS rev 14994. Phones on extensions: 1) x-lite 2) cisco sip phone 3) audio played by fs to the extension being eavesdropped I did not change any codec configuration, I just use the standard one that comes with both FS and the phones. Some time ago someone on FS irc channel told me that this is just how FS eavesdropping works... from your response I understand that this is not entirely true? Maciej Aniserowicz Anthony Minessale wrote: > > That's is a somewhat vague position. > > You did not mention which version of FreeSWITCH you are running, the > phones > being used in your example, your configuration, the codecs in use etc. > > BTW, > I think you should only ask one question at a time on this list. The list > is run by volunteers and it's sort of rude to expect 3 or 4 threads to be > tended to concerning the same one individual. > > > 2009/10/5 Maciej Aniserowicz > >> Hello, >> When I use eavesdropping in FreeSWITCH, the sound quality is really bad. >> Is >> there any way to improve it? Is this a known problem? >> Br/ >> Maciej Aniserowicz >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> 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_miness...@hotmail.com > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > iax:gu...@conference.freeswitch.org/888 > googletalk:conf+...@conference.freeswitch.org > pstn:213-799-1400 > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3780245.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Bad sound quality while eavesdropping
Hello, When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is there any way to improve it? Is this a known problem? Br/ Maciej Aniserowicz___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] gateway FS informs it's client FS about users hanged up with a long delay
Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all users
register to the 'external' instance which acts as a gateway by 'internal'
instance (which in turn is controlled by my applicaiton with commands sent by
socket).
When user hangs up, the 'hanged up' event is propagated to the 'internal'
instance after a long time (~3 minutes) instead of being propagated immediately.
What can cause this issue?
Br/
Maciej Aniserowicz___
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[Freeswitch-users] Recording creates a 388-byte long file and deletes it
Hi, When I record a call in FS, it only creates a 388-byte-long wav file. The conversation is no written there, and FS deletes the file when the session finishes. What can cause this strange behavior? Br/ Maciej Aniserowicz___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
