Re: [Freeswitch-users] How to load user account from databse ?
Hi noob and Michael, thanks for your answers, I'll try to use mod_xml_curl. Hi Henry, http://wiki.freeswitch.org/wiki/Mod_xml_curl has mentioned, FS will post a request to webserver when it get a registration request. you can refer to the doc for more detail. Lei.Tang lei.tl...@gmail.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] SIP provider with extern rtp server
I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan ___ 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] Playing Background music as well as a file
2009/10/27 lakshmanan ganapathy lakindi...@gmail.com: Hi all, I've done experimenting with the uuid_displace and mux. What mux does is playing a file when the conversation is also happening. But I've a different requirement. Is posible to use uuid_displace in dialplan ? i want to do music background base on called id I need to play a background music to a UUID, that will get played continuously and also I need to play some other voice message to that uuid. Is it possible in freeswitch? If so please guide me on how to do that! ___ 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 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] Resend: Issues with SIP + TCP?
Thanks Brian, it works now. I'll try to learn to say NO next time :) /Peter -Ursprungligt meddelande- Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 26 oktober 2009 19:40 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP? At some point we'll have to NO NO NO fix your broken crap. :P The reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with me... NO! /b On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote: I understand your frustration :) We deal with SIP integration with about 10 different PBX vendors today, And it's always something that doesn't work as it should. Right now I don't have anything more connected to FS though. /Peter ___ 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 !DSPAM:4ae5ef4a32936548940180! ___ 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] how to config FS with two net interface?
Hi all, I run FS on a machine with two net interface, each interface has a
ip addr, one of the them connect to public network(has ip addr A), the
other connect to a private network(has ip addr B), FS server as a SIP
server for public through A, all outbound call will bridge to a softswitch
in private network through B. here is my sofia config file and diaplan
config:
sofia internal.xml
param name=rtp-ip value=A/
param name=sip-ip value=A/
sofia external.xml
param name=rtp-ip value=B/
param name=sip-ip value=B/
dialplan
..
extension name=OUTBOUND
condition field=destination_number expression=^(\d+)$
action application=set data=hangup_after_bridge=true/
action application=set
data=continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION/
action application=set
data=effective_caller_id_number=xxx/ !--here change the caller
number --
action application=bridge
data=sofia/external/${destination_numb...@x/
/condition
/extension
.
then call seq is
sipAgent -- [internal --(bridge)--external] --softswith
FREESWITCH
the question is, when sipAgent make a outbound call, FS can't recevie the
caller's up audio stream, I traced the SIP packets, found that FS has return
addr B in SDP when ack the invite request from sipAgent, the ack packet is
===
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
From: 1000 sip:x...@a;tag=cb4d3c4e
To: 65960581 sip:x...@a;tag=DtvSc0QX01yKN
Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
CSeq: 2 INVITE
Contact: sip:xxx...@b:5060;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
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: 245
v=0
o=FreeSWITCH 1256598185 1256598186 IN IP4 B ;wrong this is the ip addr
of the adapter connect to the private network
s=FreeSWITCH
c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the
private network
t=0 0
m=audio 31066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
I think FS should return A in SDP, not the external binding addr (B), does
somebody known how to solve this problem?
--
Lei.Tang
lei.tl...@gmail.com
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[Freeswitch-users] reload from managed
Hi, how can i call reload from managed code? and i called Csharp_switch_xml_open_root, this is working first time, then onwards it is not working. is this the correct function to call reload? -- Srinivasula Reddy K ___ 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] switch_xml_open_root
** Hi, when i am calling switch_xml_open_root(1,err) . i am getting this warning message. HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 0016, 100E9FD0 ). can any know. please help me. Thanks Srinivasula Reddy K ___ 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] Playing Background music as well as a file
I tried the uuid_displace by using the Event Socket Outbound. Not tried within dialplan, and I'm not sure whether that could be done in dialplan itself. On Tue, Oct 27, 2009 at 12:37 PM, Dome Charoenyost d...@tel.co.th wrote: 2009/10/27 lakshmanan ganapathy lakindi...@gmail.com: Hi all, I've done experimenting with the uuid_displace and mux. What mux does is playing a file when the conversation is also happening. But I've a different requirement. Is posible to use uuid_displace in dialplan ? i want to do music background base on called id I need to play a background music to a UUID, that will get played continuously and also I need to play some other voice message to that uuid. Is it possible in freeswitch? If so please guide me on how to do that! ___ 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 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 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] how to config FS with two net interface?
Try setting ext-rtp-ip and ext-sip-ip on both profiles.
On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi all, I run FS on a machine with two net interface, each interface has a
ip addr, one of the them connect to public network(has ip addr A), the
other connect to a private network(has ip addr B), FS server as a SIP
server for public through A, all outbound call will bridge to a softswitch
in private network through B. here is my sofia config file and diaplan
config:
sofia internal.xml
param name=rtp-ip value=A/
param name=sip-ip value=A/
sofia external.xml
param name=rtp-ip value=B/
param name=sip-ip value=B/
dialplan
..
extension name=OUTBOUND
condition field=destination_number expression=^(\d+)$
action application=set data=hangup_after_bridge=true/
action application=set
data=continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION/
action application=set
data=effective_caller_id_number=xxx/ !--here change the caller
number --
action application=bridge
data=sofia/external/${destination_numb...@x/
/condition
/extension
.
then call seq is
sipAgent -- [internal --(bridge)--external] --softswith
FREESWITCH
the question is, when sipAgent make a outbound call, FS can't recevie the
caller's up audio stream, I traced the SIP packets, found that FS has return
addr B in SDP when ack the invite request from sipAgent, the ack packet is
===
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
From: 1000 sip:x...@a;tag=cb4d3c4e
To: 65960581 sip:x...@a;tag=DtvSc0QX01yKN
Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
CSeq: 2 INVITE
Contact: sip:xxx...@b:5060;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
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: 245
v=0
o=FreeSWITCH 1256598185 1256598186 IN IP4 B ;wrong this is the ip addr
of the adapter connect to the private network
s=FreeSWITCH
c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the
private network
t=0 0
m=audio 31066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
I think FS should return A in SDP, not the external binding addr (B), does
somebody known how to solve this problem?
--
Lei.Tang
lei.tl...@gmail.com
___
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--
Eliot Gable
We do not inherit the Earth from our ancestors: we borrow it from our
children. ~David Brower
I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime. ~David Brower
Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero
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Re: [Freeswitch-users] SIP provider with extern rtp server
Make sure you let their media IPs through your firewall. Also, if you are behind a NAT, check you have things passing to the correct internal address. On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold i...@myrvold.org wrote: I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan ___ 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 -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ 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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View message @
http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3895104.html
To unsubscribe from Re: Can not record session. Media not enabled on
channel., click here.
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Sent from the freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] Can not record session. Media not enabled on channel.
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!
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:* [hidden
email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3899478i=0
*To:* [hidden
email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3899478i=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=nodenode=3895104i=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
___
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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
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archivehttp://n2.nabble.com/freeswitch-users-f2379917.htmlat Nabble.com.
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FreeSWITCH
Re: [Freeswitch-users] reload from managed
Is it necessary for you to call the reload function directly? Why not make an api call using switch_api_execute and the command reloadxml. On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy srinivas.ksvre...@gmail.com wrote: Hi, how can i call reload from managed code? and i called Csharp_switch_xml_open_root, this is working first time, then onwards it is not working. is this the correct function to call reload? -- Srinivasula Reddy K ___ 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 -- -Rupa ___ 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] Problem with inbound call answered but no sound
The issue is you're not behind nat-pmp or upnp so it can't figure out what your public IP is... you'll have to fill that in manually or enable those options on your router if possible. /b On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote: I haven’t changed anything since v15183, where it worked OK. In conf/sip_profiles/external: param name=ext-rtp-ip value=auto-nat/ param name=ext-sip-ip value=auto-nat/ And ___ 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] SIP provider with extern rtp server
The server is on a public IP, so there is no nat issue here. I can also see the rtp messages on wireshark starting just after the 183 Session Progress message on the server, but just in one direction, coming in to the server. So it looks like Freeswitch is stopping the rtp. Is this because the rtp originates from another ip than the sip provider ip? Ivan Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable: Make sure you let their media IPs through your firewall. Also, if you are behind a NAT, check you have things passing to the correct internal address. On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold i...@myrvold.org wrote: I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan ___ 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 -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ 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 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] Problem with inbound call answered but no sound
Thanks for the reply, Brian. Did something in FS change between v15183 and v15225 to make this occur? I ask because this same configuration worked OK in the earlier version. Lars From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, October 27, 2009 8:26 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem with inbound call answered but no sound The issue is you're not behind nat-pmp or upnp so it can't figure out what your public IP is... you'll have to fill that in manually or enable those options on your router if possible. /b On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote: I haven't changed anything since v15183, where it worked OK. In conf/sip_profiles/external: param name=ext-rtp-ip value=auto-nat/ param name=ext-sip-ip value=auto-nat/ And ___ 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] Problem with inbound call answered but no sound
Let me review that and see I can't off hand think of anything that would cause that. /b On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote: Thanks for the reply, Brian. Did something in FS change between v15183 and v15225 to make this occur? I ask because this same configuration worked OK in the earlier version. Lars ___ 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] SIP provider with extern rtp server
No, the IP address the media originates from does not need to be tied to the SIP IP address. Can you send a Wireshark capture taken on the FreeSWITCH server of both call legs? Or, if you can, pastebin a debug log from FreeSWITCH console with sofia loglevel set to 9 and siptrace on for any Sofia SIP profiles involved. On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold i...@myrvold.org wrote: The server is on a public IP, so there is no nat issue here. I can also see the rtp messages on wireshark starting just after the 183 Session Progress message on the server, but just in one direction, coming in to the server. So it looks like Freeswitch is stopping the rtp. Is this because the rtp originates from another ip than the sip provider ip? Ivan Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable: Make sure you let their media IPs through your firewall. Also, if you are behind a NAT, check you have things passing to the correct internal address. On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold i...@myrvold.org wrote: I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan ___ 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 -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ 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 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 -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ 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] Forcing endpoint registration
Is the following the correct command to force registration of an endpoint below which is not showing up in 'sofia status profile internal' currently, but when it does it looks like the following? Call-ID:mjkg8c3zlqwjzfd.hrt5 User: 1...@192.168.10.29 Contact:user sip:1...@192.168.10.103;line=3852 Agent: snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1) Status: Registered(UDP)(unknown) EXP(2009-10-27 10:46:07) Host: fs IP: 192.168.10.103 Port: 5060 Auth-User: 1019 Auth-Realm: 192.168.10.29 MWI-Account:1...@192.168.10.29 sofia profile internal flush_inbound_reg mailto:1...@192.168.10.29 1...@192.168.10.29 Thanks Lars ___ 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] FreeSWITCH Weekly Conference Call Agenda - Oct 30th
Greetings all! I just wanted to say thank you to those who have been contributing to the documentation and cleanup efforts. Please keep up the good work. FYI, we still have lots of janitorial stuff to do. Just keep checking the latest conf call agenda for updates. This week's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30 We are always looking for people to help out with documentation and other little projects so please feel free to ask me if you would like to assist. Also, we are working on having members of the community give brief presentations during the weekly conference calls. We want people to join the conference and share with the group information about how they're using FreeSWITCH in production, or they can give little tutorials on how to use various parts of FS. We are also working on getting a web-cast enabled so that we can have a little video to go with the audio. If you have something you'd like to discuss with group please let me know. Thanks for being such a great community! -Michael ___ 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] Forcing endpoint registration
From FreeSWITCH you can't force and endpoint to register... thats the endpoints job... what you're doing is just flushing the reg out of FreeSWITCH so now it can't find the endpoint... Try adding a reboot on there maybe i'll reboot the device and cause it to register again... /b On Oct 27, 2009, at 12:33 PM, Lars Zeb wrote: Is the following the correct command to force registration of an endpoint below which is not showing up in ‘sofia status profile internal’ currently, but when it does it looks like the following? Call-ID:mjkg8c3zlqwjzfd.hrt5 User: 1...@192.168.10.29 Contact:user sip:1...@192.168.10.103;line=3852 Agent: snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1) Status: Registered(UDP)(unknown) EXP(2009-10-27 10:46:07) Host: fs IP: 192.168.10.103 Port: 5060 Auth-User: 1019 Auth-Realm: 192.168.10.29 MWI-Account:1...@192.168.10.29 sofia profile internal flush_inbound_reg 1...@192.168.10.29 Thanks Lars ___ 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] FS Training
Did the voting booth close? I was unable to vote. I'm not sure what link to click and I have had some strange issues with my FS account today. I would be interested in paid training. Do you have plans for offering a training session at your locale? Or would you travel onsite to provide training? Best Regards, Jerry ___ 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] Estimating Call Capacity
A little off-topic, but since call-capacity is the subject, what are people using to analyze their CDR's to discover this? I'm handling about 30k calls per day but have only a bandwidth-based guesstimate of the peak number of concurrent calls I'm handling. If there's an open source solution, I'd appreciate a pointer. Regards, Cliff On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote: Although, FYI, I just benchmarked mod_xml_curl on a separate web app server from FS with FS on a Dell R710 with their current best processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32 GB memory. The web app server is less than half the power of the R710. I maxed the web app server at 300 calls per second (both setting up and tearing down) and the R710 running FS was 65% idle. No audio was being proxied through FS, though. If I were running the web app server on an equivalent R710, they probably would have been on-par with each other in performance. Extrapolating, I expect that in such a case I should be able to get at least 650 CPS out of FS, though for production I would probably limit it to 400 CPS or less so I leave room for miscellaneous tasks. I maxed out the R710 at over 16,000 simultaneous calls (again, no audio proxying) but the only reason I couldn't do more was because I hit some sort of thread creation limit in Linux. There was about 17 GB of memory used for this many calls. This should give you some ballpark idea of what you can accomplish with FS. At some point, I will track down and resolve the thread creation issue, at which time I believe call limits will be limited either by a complex combination of available memory, the speed of the processor, the cost of thread context switching, calls per second setup rate, and call duration. -- Eliot Gable -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, October 26, 2009 4:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware. Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ 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 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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 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 -- http://www.google.com/search?q=vonage+sucks ___ 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] Estimating Call Capacity
I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E. you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs the mirrors, tracking all sorts of good data. You can install it on a Centos box, and get a free trial. http://www.manageengine.com/products/vqmanager/index.html What is really cool is that it actually monitors the RTP/RTCP as well as all of the SIP headers and archives the calls, so you can look at calls from several days ago and see EXACTLY what happened on them. I have used this extensively to pinpoint bad Level 3 and X/O media gateways.. Much better than trying to sniff packets in real-time and MAYBE catch a problem.. I've also used it to find/fix several SIP issues w/ odd endpoints.. Very easy to see.. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch- users-boun...@lists.freeswitch.org] On Behalf Of Cliff Wells Sent: Tuesday, October 27, 2009 3:52 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity A little off-topic, but since call-capacity is the subject, what are people using to analyze their CDR's to discover this? I'm handling about 30k calls per day but have only a bandwidth-based guesstimate of the peak number of concurrent calls I'm handling. If there's an open source solution, I'd appreciate a pointer. Regards, Cliff On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote: Although, FYI, I just benchmarked mod_xml_curl on a separate web app server from FS with FS on a Dell R710 with their current best processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32 GB memory. The web app server is less than half the power of the R710. I maxed the web app server at 300 calls per second (both setting up and tearing down) and the R710 running FS was 65% idle. No audio was being proxied through FS, though. If I were running the web app server on an equivalent R710, they probably would have been on-par with each other in performance. Extrapolating, I expect that in such a case I should be able to get at least 650 CPS out of FS, though for production I would probably limit it to 400 CPS or less so I leave room for miscellaneous tasks. I maxed out the R710 at over 16,000 simultaneous calls (again, no audio proxying) but the only reason I couldn't do more was because I hit some sort of thread creation limit in Linux. There was about 17 GB of memory used for this many calls. This should give you some ballpark idea of what you can accomplish with FS. At some point, I will track down and resolve the thread creation issue, at which time I believe call limits will be limited either by a complex combination of available memory, the speed of the processor, the cost of thread context switching, calls per second setup rate, and call duration. -- Eliot Gable -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, October 26, 2009 4:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+ x2200+M2 Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware. Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ 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 mailing list
Re: [Freeswitch-users] Estimating Call Capacity
Cliff, Try using xml_rpc ... status or show channels will give you what you need. SDR Cliff Wells wrote: A little off-topic, but since call-capacity is the subject, what are people using to analyze their CDR's to discover this? I'm handling about 30k calls per day but have only a bandwidth-based guesstimate of the peak number of concurrent calls I'm handling. If there's an open source solution, I'd appreciate a pointer. Regards, Cliff On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote: Although, FYI, I just benchmarked mod_xml_curl on a separate web app server from FS with FS on a Dell R710 with their current best processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32 GB memory. The web app server is less than half the power of the R710. I maxed the web app server at 300 calls per second (both setting up and tearing down) and the R710 running FS was 65% idle. No audio was being proxied through FS, though. If I were running the web app server on an equivalent R710, they probably would have been on-par with each other in performance. Extrapolating, I expect that in such a case I should be able to get at least 650 CPS out of FS, though for production I would probably limit it to 400 CPS or less so I leave room for miscellaneous tasks. I maxed out the R710 at over 16,000 simultaneous calls (again, no audio proxying) but the only reason I couldn't do more was because I hit some sort of thread creation limit in Linux. There was about 17 GB of memory used for this many calls. This should give you some ballpark idea of what you can accomplish with FS. At some point, I will track down and resolve the thread creation issue, at which time I believe call limits will be limited either by a complex combination of available memory, the speed of the processor, the cost of thread context switching, calls per second setup rate, and call duration. -- Eliot Gable -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, October 26, 2009 4:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware. Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ 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 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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 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 mailing list FreeSWITCH-users@lists.freeswitch.org
Re: [Freeswitch-users] FS Training
On Tue, Oct 27, 2009 at 12:17 PM, Jerry Richards jerry.richa...@teotech.com wrote: Did the voting booth close? I was unable to vote. I'm not sure what link to click and I have had some strange issues with my FS account today. I would be interested in paid training. Do you have plans for offering a training session at your locale? Or would you travel onsite to provide training? Jerry, Yes the poll closed last Thursday. I'm glad to hear that you are interested in paid training. We are still evaluating all the options. I will be contacting people off list to discuss specifics. This is still in the incubation stage, so no ETA on anything just yet. -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
Re: [Freeswitch-users] switch_xml_open_root
You probably should not be calling that function, what are you trying to do? Mike On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote: Hi, when i am calling switch_xml_open_root(1,err) . i am getting this warning message. HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap ( 0016, 100E9FD0 ). can any know. please help me. ___ 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] switch_xml_open_root
On Tue, Oct 27, 2009 at 2:31 PM, Michael Jerris m...@jerris.com wrote: You probably should not be calling that function, what are you trying to do? Mike On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote: You might also want to move this conversation to the -dev list because it's a little intense for the -users list. -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
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-23 15:14 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TC TC so the real solution is to implement a check for CallProceeding , Progress TCand Facility message whether it has a faststart element included. It it is TCtrue than you might start pre_answer. TC TC TCalso, i don't see any handling of Call Proceeding ... what if there is a TCfastStart element in CallProceeding message? :) Handling of fastStart in CallProceeding is commented out in h323plus library, this is exploration from h323plus developers about this: Yes that should be mera. The problem is that Callproceeding does not always come from the remote it may be generated by the gatekeeper. MERA where sending fast start elements in the Call proceeding and connect. The call proceeding where not valid and causing the media to fail. Normally (although valid) EP's do not set Fast Start in Call proceeding so the code was disabled to resolve the MERA issue. if you wont read bugs file in mod_h323, there is explaned how to enable it. C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ 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] how to config FS with two net interface?
Thanks Eliot, It works.
2009/10/27 Eliot Gable
egable+freeswi...@gmail.comegable%2bfreeswi...@gmail.com
Try setting ext-rtp-ip and ext-sip-ip on both profiles.
On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi all, I run FS on a machine with two net interface, each interface has
a
ip addr, one of the them connect to public network(has ip addr A), the
other connect to a private network(has ip addr B), FS server as a SIP
server for public through A, all outbound call will bridge to a
softswitch
in private network through B. here is my sofia config file and diaplan
config:
sofia internal.xml
param name=rtp-ip value=A/
param name=sip-ip value=A/
sofia external.xml
param name=rtp-ip value=B/
param name=sip-ip value=B/
dialplan
..
extension name=OUTBOUND
condition field=destination_number expression=^(\d+)$
action application=set data=hangup_after_bridge=true/
action application=set
data=continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION/
action application=set
data=effective_caller_id_number=xxx/ !--here change the caller
number --
action application=bridge
data=sofia/external/${destination_numb...@x/
/condition
/extension
.
then call seq is
sipAgent -- [internal --(bridge)--external] --softswith
FREESWITCH
the question is, when sipAgent make a outbound call, FS can't recevie the
caller's up audio stream, I traced the SIP packets, found that FS has
return
addr B in SDP when ack the invite request from sipAgent, the ack packet
is
===
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
From: 1000 sip:x...@a;tag=cb4d3c4e
To: 65960581 sip:x...@a;tag=DtvSc0QX01yKN
Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
CSeq: 2 INVITE
Contact: sip:xxx...@b:5060;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
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: 245
v=0
o=FreeSWITCH 1256598185 1256598186 IN IP4 B ;wrong this is the ip
addr
of the adapter connect to the private network
s=FreeSWITCH
c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the
private network
t=0 0
m=audio 31066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
I think FS should return A in SDP, not the external binding addr (B),
does
somebody known how to solve this problem?
--
Lei.Tang
lei.tl...@gmail.com
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--
Eliot Gable
We do not inherit the Earth from our ancestors: we borrow it from our
children. ~David Brower
I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime. ~David Brower
Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero
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Lei.Tang
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Re: [Freeswitch-users] how to config FS with two net interface?
you should have just needed to set the rtp-ip and the sip-ip on both profiles to their values and it would have worked fine... their would have been no need to set the ext-*-ip equiv. /b On Oct 27, 2009, at 8:14 PM, Lei Tang wrote: Thanks Eliot, It works. ___ 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] how to config FS with two net interface?
Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when I set the ext-rtp-ip, it works fine. 2009/10/28 Brian West br...@freeswitch.org you should have just needed to set the rtp-ip and the sip-ip on both profiles to their values and it would have worked fine... their would have been no need to set the ext-*-ip equiv. /b On Oct 27, 2009, at 8:14 PM, Lei Tang wrote: Thanks Eliot, It works. ___ 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 -- Lei.Tang lei.tl...@gmail.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] how to config FS with two net interface?
Ok then something is broken badly... and that makes NO sense. Because it works on my box. /b On Oct 27, 2009, at 8:34 PM, Lei Tang wrote: Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when I set the ext-rtp-ip, it works fine. 2009/10/28 Brian West br...@freeswitch.org you should have just needed to set the rtp-ip and the sip-ip on both profiles to their values and it would have worked fine... their would have been no need to set the ext-*-ip equiv. /b On Oct 27, 2009, at 8:14 PM, Lei Tang wrote: Thanks Eliot, It works. ___ 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 -- Lei.Tang lei.tl...@gmail.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 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] switch_xml_open_root
Hi mike, thank for your reply. i am trying to call that function from swig.cs. its working fine first time with the warning information, then onwards it is not working. is there other way can i call reload function from freeswitch.managed code. any help srinivas On Wed, Oct 28, 2009 at 3:01 AM, Michael Jerris m...@jerris.com wrote: You probably should not be calling that function, what are you trying to do? Mike On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote: * * Hi, when i am calling switch_xml_open_root(1,err) . i am getting this warning message. HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 0016, 100E9FD0 ). can any know. please help me. ___ 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 -- Srinivasula Reddy K ___ 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] Retrieve conference member state using cli?
Just like you can call conference list for conferences, is there a way to retrieve the profile state of a conference member using the cli or xml/rpc? Lon ___ 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] switch_xml_open_root
just api execute reloadxml Mike On Oct 28, 2009, at 12:37 AM, srinivasula reddy wrote: Hi mike, thank for your reply. i am trying to call that function from swig.cs. its working fine first time with the warning information, then onwards it is not working. is there other way can i call reload function from freeswitch.managed code. any help ___ 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] reload from managed
i tried switch_api_execute its working fine, thanks rupa On Tue, Oct 27, 2009 at 8:48 PM, Rupa Schomaker r...@rupa.com wrote: Is it necessary for you to call the reload function directly? Why not make an api call using switch_api_execute and the command reloadxml. On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy srinivas.ksvre...@gmail.com wrote: Hi, how can i call reload from managed code? and i called Csharp_switch_xml_open_root, this is working first time, then onwards it is not working. is this the correct function to call reload? -- Srinivasula Reddy K ___ 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 -- -Rupa ___ 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 -- Srinivasula Reddy K ___ 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
