Re: [Freeswitch-users] Delay when transferring call
On Fri, Sep 18, 2009 at 10:41:19AM +0200, Sias Mey wrote: Indeed the debug info did shed some light. I know at least one other poster has asked about that before. It seems to be an error related to accesing mysql InnoDB via odbc. Something about not decreasing the thread count before closing the connection. Unfortunately the only information I can get on fixing the bug says replace you PHP mysql.dll with and older version and restart IIS??? Not so usefull in my case of running in linux and not using PHP. I have tried adding a db.close to all my scripts but that doesent seem to help either. So, if any of the rest of you have come across this and know how to fix in if im not using php and IIS, some help would be greatly appreciated. I think in the mean time I will try to recompile my mysql odbc lib (I installed from the package manager initially). Im not going to post a bug to freeswitch since this seems to be a mysql odbc related issue, but thankyou for the help in tracking it down to that. Hi, after some more googleing I finally found a mysql forum post claiming this was fixed in the 3.51.23 version of the mysql odbc connector. Suprise suprise packaged with ubuntu comes 3.51.17 Needless to say a rather messy recomile of a whole bunch of tool finally got me just the new odbc driver and that seems to have solved it. Thank you again for your help. You guys are great at helping even if the problem isent with your app. Cheers, Sias ___ 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] conference participant from behind NAT
I am a bit confused with what's going on in a following scenario. I have a public FS server with a public conference, that clients are connecting to with my softphone. All of this softphones have STUN option enabled and working, effectively resolving client's public IP address. They also have ICE enabled (but I guess it's not relevant here, since FS doesn't do ICE). Also, media trafic is secured with SRTP. The problem is when one client connects from port-restricted NAT into a conference he can hear sound for some time and he can be heard by other participants, but after awhile sound is gone and neither he hear anything nor he can be heard. Where is the problem? Is it NAT, closing RTP port after some silence period from client? I tried to start conference with waste flag, but without success eventually. The very same person can be contacted through this FS with direct call (being established in proxy_media mode) without any problems, but this is where ICE stuff starts doing its' magic, I guess. Maybe I should try the same with SRTP disabled? Any help would be apreciated! Best regards, Robert. ___ 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] conference participant from behind NAT
RobertT siniy...@gmail.com wrote: Where is the problem? Is it NAT, closing RTP port after some silence period from client? It could be a time-out, i.e., the nat router isn't keeping the port translation alive. I don't like nat at all. As more people migrate to IPv6 the problem will gradually go away. ___ 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] conference participant from behind NAT
Are there ways to escape this timeouts exchanging RTP with FS? Why didn't waste flag help? Maybe I should flood channel in both directions? Will CNG on a client side be a good descision? =) ___ 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] [ERR] mod_dingaling.c:980 Stun Failed!
Hi! I have just started to use dingaling again, and noticed I constantly get a stun error. 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed! stun.fwdnet.net:3478 [Remote Address Error!] I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers and keep getting this error with dingaling. I have no problems with inbound sip calls, so I don't think its the actual stun server. Has anyone else seen this? I am using: FreeSWITCH Version 1.0.trunk (14952) Thanks! ___ 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] Dynamic volume adjustment in heterogenous IVR menus
Sorry for the email subject that sounds like a IEEE paper. I am building IVRs using FS API and sending out audio that is a combination of TTS and playing WAV files. What is the best way to control volume levels? I know i might be asking for magic here... In any case, is there any simple ways to add gain to certain nodes of am IVR? /aep -- Stopping junk mailers is good for the environment ___ 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] Attended transfer - no audio
I commented out the following in our internal profile: param name=media-option value=bypass-media-after-att-xfer/ Which helped. We don't use bypass media though.. Jan On Wed, Sep 23, 2009 at 8:59 AM, Michael Jerris m...@jerris.com wrote: Did you ever resolve this issue? If not, please make sure you open a bug on jira.freeswitch.org with as much detail to reproduce this as possible. Mike On Sep 10, 2009, at 6:14 PM, Jan Kubr wrote: Hi, we have a Freeswitch server on a public IP and a few phones behind NAT. The phones are configured to use STUN and can register and call each other fine. The problem is that after attended transfer (using the mechanism the phones provide - REFER) is finished, the two parties can't hear each other. This problem doesn't occur when the phones are in the same subnet as Freeswitch. I know this isn't enough information to solve the problem, but do you have any hints on how to debug this? Are there any specific Freeswitch settings that could help us? Thanks, Jan ___ 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] Proxy|Bypass Media Wrong Payload.
NO, proxy media and bypass media are wildly different behaviors and do process things a little differently. /b On Sep 29, 2009, at 12:40 AM, Mariano de Llano wrote: it seams very weird to me that Sofia uses different approach to parse the mappings depending if it is handling or not the media (Perhaps is ignoring it and using a correct one). ___ 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] Proxy|Bypass Media Wrong Payload.
Type 'sofia loglevel all 9' then 'sofia profile siptrace on' replace with the profile name then press F8 to turn debug log on Capture the whole thing and email me the log. I can pretty much tell you sofia is pissed about something in the SDP but I wanna see the logs. Thanks, Brian On Sep 29, 2009, at 12:40 AM, Mariano de Llano wrote: What do you suggest me to do? (Use a hammer with my 3K AddPacs it's not an option) :D ___ 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] Attended transfer - no audio
NAT involved? /b On Sep 29, 2009, at 7:27 AM, Jan Kubr wrote: I commented out the following in our internal profile: param name=media-option value=bypass-media-after-att-xfer/ Which helped. We don't use bypass media though.. Jan ___ 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] conference participant from behind NAT
ты все еще наблюдаешь эту проблему? я думал она уже решена... эни вей, я уже приехал и сделаю скоро воторой IP нам для собственного STUN-сервера. -- Best regards, Dmitry Kadantsev http://www.doxwox.com - Best web meeting and online collaboration tool. On Tue, Sep 29, 2009 at 10:32 AM, RobertT siniy...@gmail.com wrote: I am a bit confused with what's going on in a following scenario. I have a public FS server with a public conference, that clients are connecting to with my softphone. All of this softphones have STUN option enabled and working, effectively resolving client's public IP address. They also have ICE enabled (but I guess it's not relevant here, since FS doesn't do ICE). Also, media trafic is secured with SRTP. The problem is when one client connects from port-restricted NAT into a conference he can hear sound for some time and he can be heard by other participants, but after awhile sound is gone and neither he hear anything nor he can be heard. Where is the problem? Is it NAT, closing RTP port after some silence period from client? I tried to start conference with waste flag, but without success eventually. The very same person can be contacted through this FS with direct call (being established in proxy_media mode) without any problems, but this is where ICE stuff starts doing its' magic, I guess. Maybe I should try the same with SRTP disabled? Any help would be apreciated! Best regards, Robert. ___ 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] Freeswitch Failover
Hi All, please let me know implementation of failover in freeswitch. Thanks Srinivas ___ 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] conference participant from behind NAT
в том то все и дело что с тобой мы эту проблему вроде как решили, а у Юры ее никогда не было. и тут на тебе... ___ 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] Alias user mapping
Hi everybody,
I have a problem with Alphanumeric to numeric user mapping
I have done like it's written here :
Alphanumeric to numeric user mapping
Say you want a user's id to be alphanumeric (like an email username), such
as johnsm...@pbx.example.com. These users have alphanumeric usernames in
their sip phone config, but you want to map them from their sip username to
a numeric extension, and vice versa.
As of version 1.0.4, Freeswitch makes this trivial to accomplish. A user's
ID can be any alphanumeric string, and this can be simply tied to an
extension number using the 'number-alias' property. This property creates an
aliased directory entry that points to the alphanumeric user entry.
*NOTE: When using this attribute, you must be careful not to create a
directory collision by having another user whose ID is the same as another
user's alias*
Here is an example from the user directory:
user id=johnsmith number-alias=1001
!-- Insert the usual user configuration variables and params here,
including user password, voicemail password, caller ID info, etc --
/user
So when a user dials extension number 1001, your dialplan can use the
'user_data' function to look up the ID attribute associated with that number
alias. In the default dialplan, the 'Local Extension' section can be made to
work with a small change to the 'bridge' line:
action application=bridge
data=user/${user_data(${dialed_extensi...@${domain_name} attr
id)}...@${domain_name}/
*NOTE: Using this user_data function in combination with mod_xml_curl will
generate an additional request each time the user_data function is called.
Note that it is already called once in the Local Extension section to
determine the callgroup. Beware of performance implications of this with
high-volume systems.*
But when I want to call my alias-number, FS says No Route, Abording
My version of FreeSWITCH is the 1.0.4pre9.
Do you have any ideas ? Did I forget something to do?
Thanks
--
Jonathan BAROU
SQLI LYON - CRCI
jba...@sqli.com
lyon.c...@sqli.com
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Re: [Freeswitch-users] Dynamic volume adjustment in heterogenous IVR menus
The best way is to start with normalized sound files, then to use whatever features are available in your tts engine to send the right volume matched to the sound files. That being said, a new media bug was just added in trunk for auto gain control and that might help, but I would never use it for this purpose. Mike On Sep 29, 2009, at 7:29 AM, Alberto Escudero wrote: Sorry for the email subject that sounds like a IEEE paper. I am building IVRs using FS API and sending out audio that is a combination of TTS and playing WAV files. What is the best way to control volume levels? I know i might be asking for magic here... In any case, is there any simple ways to add gain to certain nodes of am IVR? ___ 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] Alias user mapping
Could you test this in svn trunk please. Mike On Sep 29, 2009, at 9:33 AM, Jonathan Barou wrote: Hi everybody, I have a problem with Alphanumeric to numeric user mapping I have done like it's written here : ... But when I want to call my alias-number, FS says No Route, Abording My version of FreeSWITCH is the 1.0.4pre9. Do you have any ideas ? Did I forget something to do? Thanks -- Jonathan BAROU SQLI LYON - CRCI jba...@sqli.com lyon.c...@sqli.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] conference participant from behind NAT
Most likely the client NAT is cutting off the translation due to no traffic. This could be because the client is not sending any traffic, regardless of settings you set on FreeSWITCH. Try disabling all vad and dtx on your soft phone to see if this helps. Also, your email seems to indicate that you have solved the problem for yourself and others have not had the problem. Is anyone still experiencing this issue? Mike ___ 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] Proxy|Bypass Media Wrong Payload.
I'm sure of that. But I was talking about how is handle the parse of the packet not the process, and also I was referring to the case when FS is actually handling the media (proxy-media=false bypass_media=false) as I said before FS ignores the las parameter in the rtrpmap when is handling the media, so, I'm quite sure that something can be done in order to make it work when is not handling it. I understand that is not a FS bug, however since it have different behavior depending on the media mode something is not working properly. On 29/09/2009, at 10:14, Brian West wrote: NO, proxy media and bypass media are wildly different behaviors and do process things a little differently. /b On Sep 29, 2009, at 12:40 AM, Mariano de Llano wrote: it seams very weird to me that Sofia uses different approach to parse the mappings depending if it is handling or not the media (Perhaps is ignoring it and using a correct one). ___ 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] Load-Balance FreeSWITCH with Ultramonkey
On Mon, Sep 28, 2009 at 9:05 PM, Even André Fiskvik grev...@me.com wrote: From: Even André Fiskvik grev...@me.com To: freeswitch-users@lists.freeswitch.org Date: Mon, 28 Sep 2009 22:52:13 +0200 Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey I have been working with a similar setup myself, but for some reason I ended up ditching theUltraMonkey setup because I just couldn't get it to work right. It's been quite a while since my effort, so I don't remember what the exact issue was. I got registrations to work, but had some other sip-dialog issues. We have since then changed over to running OpenSIPs as a loadbalancer in front of multiple FreeSWITCH instances. This setup is still in testing, but seemlingy works fine (and if it doesn't, it's my own fault for writing a bad opensips config). After we have done some more testing I can create a wiki-page with config details. Best regards, Even André Thanks, Even, that would be great! I might have to give up on the ultramonkey solution, since I can't find anyone who has made it work. It's too bad, because it would fit well with the rest of our architecture. Mike van Lammeren ___ 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] Alias user mapping
I'm sorry but I'm new in the freeswitch communauty, what I have to do to test this in svn trunk ? Thanks 2009/9/29 Michael Jerris m...@jerris.com Could you test this in svn trunk please. Mike On Sep 29, 2009, at 9:33 AM, Jonathan Barou wrote: Hi everybody, I have a problem with Alphanumeric to numeric user mapping I have done like it's written here : ... But when I want to call my alias-number, FS says No Route, Abording My version of FreeSWITCH is the 1.0.4pre9. Do you have any ideas ? Did I forget something to do? Thanks -- Jonathan BAROU SQLI LYON - CRCI jba...@sqli.com lyon.c...@sqli.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 -- Jonathan BAROU SQLI LYON - CRCI jba...@sqli.com lyon.c...@sqli.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] Alias user mapping
http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code http://wiki.freeswitch.org/wiki/Installation_Guide#Compiling_the_Source_Code On Sep 29, 2009, at 10:19 AM, Jonathan Barou wrote: I'm sorry but I'm new in the freeswitch communauty, what I have to do to test this in svn trunk ? ___ 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] Freeswitch Failover
On 29/09/09 16:37 +0530, srinivasula reddy wrote: Hi All, please let me know implementation of failover in freeswitch. I'm also interested in this topic. Obviously there are some things which are harder to failover than others, like event socket connections, or agent queue states, but what about registrations and A-leg + B-leg type calls, where the media does not flow through FreeSwitch? Are there mechanisms for 'loading' a list of registrations or call states into a FreeSwitch box? -- Dan White ___ 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] Transfer oddity
Has anyone given this any thought? Do I need to provide more information?
It's still not making any sense to me, and I'm planning on just
removing all of the default dialplans, but I'd like to make sure this
won't recur in the future.
BB
On Fri, Sep 25, 2009 at 9:33 AM, Bradley Brashier bjbrash...@gmail.com wrote:
Hi guys,
I've got a strange situation that I'm at a loss to explain. With all
callers, I go through a dialplan where I check to see if they should
be a moderator, then transfer them to another which puts them into a
conference accordingly. This worked great on one server, but when I
copied the code to another server (both running CentOS), the transfer
no longer works properly. Here's a log snippet from the incorrectly
working server:
Dialplan: sofia/internal/14258291...@10.10.67.190 Regex (FAIL)
[hold_music] destination_number(7001) =~ /^$/ break=on-false
Dialplan: sofia/internal/14258291...@10.10.67.190 Regex (FAIL)
[hold_music] ${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=on-false
Dialplan: sofia/internal/14258291...@10.10.67.190 ANTI-Action
set(zrtp_enrollment=true)
Dialplan: sofia/internal/14258291...@10.10.67.190 ANTI-Action answer()
Dialplan: sofia/internal/14258291...@10.10.67.190 ANTI-Action
playback(/usr/local/freeswitch/sounds/vpbx/moh.wav)
2009-09-25 07:51:31.204920 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/14258291...@10.10.67.190) State Change CS_ROUTING -
CS_EXECUTE
2
(note that the 7001 in the first line is the number I chose for my dialplan)
On the working server, the first line is still there, but the second
(and further) is replaced by further checks to see if it might be my
conference dialplans, which is what I would expect. I looked into
dialplans/default.xml, and the code for the above is there, but let me
copy it here again to discuss:
extension name=hold_music
condition field=destination_number expression=^$/
condition field=${sip_has_crypto}
expression=^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$
action application=answer/
action application=execute_extension data=is_secure XML features/
action application=playback data=$${hold_music}/
!-- This really should be an IVR for zrtp enrollment but this
is just a demo--
anti-action application=set data=zrtp_enrollment=true/
anti-action application=answer/
anti-action application=playback data=$${hold_music}/
/condition
/extension
Now, the way I understand this, it says that if the number is , it
should check the 2nd condition (which says to play hold music in a
couple of different flavors), but if the number is NOT , it should
go past, not even checking the 2nd condition. This understanding is
corroborated by the working server, which does indeed skip past and
not check the 2nd condition. Does anyone know why a server might be
going into a conditional that it knows it failed on?
For what it's worth, both servers are running on the current trunk,
with the only change being the addition of flite.
BB
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[Freeswitch-users] Loop detection in dialplan?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, is it possible to detect and avoid loops in dialplan caused by two or more extensions which create a redirect chain? regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKwiyk4tZeNddg3dwRAjA9AJ4gMsNNHgs0/FdzHY9aR0w/ftflygCfWU7F IpADQ02MuHm5WsecOOeU2DU= =JjI9 -END PGP SIGNATURE- ___ 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] More than One DID
Say, could someone please direct me to information on registering more than one DID with a SIP provider? When I try this using two XML files in ~/conf/sip_profiles/external, I find only one or the other registers (both work fine when I use them separately). I don't think the issue is with the provider as we had both DIDs working on our old VoIP server. I'm using the same info. I'm still pretty new, so I suspect the answer is in front of my face. I could use a nudge. Regards, Mike G. ___ 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] More than One DID
On Tue, Sep 29, Michael Gende wrote: Say, could someone please direct me to information on registering more than one DID with a SIP provider? When I try this using two XML files in ~/conf/sip_profiles/external, I find only one or the other registers (both work fine when I use them separately). Please post your configs with out passwords. Thanks. --FC ___ 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] Searching Mailing List Archives
Sorry for this mundane question, but how do I search mailing archives for keywords? The following link has no search option? http://lists.freeswitch.org/pipermail/freeswitch-users/ Thanks And 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] More than One DID
Here they are, mildly modified: ~/conf/sip_profiles/external/gateway1212.xml (name changed): include gateway name=33.44.55.66 param name=username value=8158381212/ param name=password value=bloodyblahbloodyblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include ~/conf/sip_profiles/external/gateway1234.xml (name changed for pseudo tn): include gateway name=33.44.55.66 param name=username value=8158381234/ param name=password value=blahbloodyblahblahblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include The gateway name is the same for both. Note that each registers and passes calls both ways on their own. I'd like to use 'em at the same time. Thanks for having a look. Mike G. On Tue, Sep 29, 2009 at 11:21 AM, Frank Carmickle fr...@carmickle.comwrote: On Tue, Sep 29, Michael Gende wrote: Say, could someone please direct me to information on registering more than one DID with a SIP provider? When I try this using two XML files in ~/conf/sip_profiles/external, I find only one or the other registers (both work fine when I use them separately). Please post your configs with out passwords. Thanks. --FC ___ 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] Searching Mailing List Archives
On Tue, Sep 29, Jerry Richards wrote: Sorry for this mundane question, but how do I search mailing archives for keywords? The following link has no search option? You can use google. site:lists.freeswitch.org yourterm --FC ___ 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] Loop detection in dialplan?
we decrement max forwards across a bridge and on transfer, so they are supposed to sort themselves out automatically, this of course won't resolve situations like loops via a provider or pstn that do not pass along max forwards. Mike On Sep 29, 2009, at 11:49 AM, Helmut Kuper wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, is it possible to detect and avoid loops in dialplan caused by two or more extensions which create a redirect chain? regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKwiyk4tZeNddg3dwRAjA9AJ4gMsNNHgs0/FdzHY9aR0w/ftflygCfWU7F IpADQ02MuHm5WsecOOeU2DU= =JjI9 -END PGP SIGNATURE- ___ 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] Searching Mailing List Archives
google site:http://lists.freeswitch.org/pipermail/freeswitch-users/ my search term here, or try nabble. Mike On Sep 29, 2009, at 12:35 PM, Jerry Richards wrote: Sorry for this mundane question, but how do I search mailing archives for keywords? The following link has no search option? http://lists.freeswitch.org/pipermail/freeswitch-users/ Thanks And 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
[Freeswitch-users] Multiple Subnets
Hello, I have some problems running a FreeSWITCH with endpoints located in different subnets. For example a FS is listening at 192.168.50.14/32 and endpoints from the same (192.168.50.0/24) subnet work as expected. But when I try to receive a media from an endpoint located at different subnet, let's say 192.168.60.0/24 the RTP stream cannot be bridged. FS sends an INVITE message asking endpoint to send media to external interface. Can anyone give me a hint where to look on how to overcome such behavior? Kind Regards, Andrey Nepomnyaschih ___ 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] More than One DID
On Tue, Sep 29, Michael Gende wrote: Here they are, mildly modified: ~/conf/sip_profiles/external/gateway1212.xml (name changed): include gateway name=33.44.55.66 param name=username value=8158381212/ param name=password value=bloodyblahbloodyblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include ~/conf/sip_profiles/external/gateway1234.xml (name changed for pseudo tn): include gateway name=33.44.55.66 param name=username value=8158381234/ param name=password value=blahbloodyblahblahblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include The gateway name is the same for both. Note that each registers and passes calls both ways on their own. I'd like to use 'em at the same time. This is what I was suspecting. You can't have two gateways with the same name. If you need to have them be the same domain set that with the option param name=realm value=33.44.55.66/ HTH --FC ___ 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] Multiple Subnets
Maybe your router isn't really a router and is doing NAT behind NAT? Need logs and sip traces because we would only be guessing at this point. /b On Sep 29, 2009, at 10:52 AM, Andrey Nepomnyaschih wrote: Hello, I have some problems running a FreeSWITCH with endpoints located in different subnets. For example a FS is listening at 192.168.50.14/32 and endpoints from the same (192.168.50.0/24) subnet work as expected. But when I try to receive a media from an endpoint located at different subnet, let’s say 192.168.60.0/24 the RTP stream cannot be bridged. FS sends an INVITE message asking endpoint to send media to external interface. Can anyone give me a hint where to look on how to overcome such behavior? Kind Regards, Andrey Nepomnyaschih ___ 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] More than One DID
Hi Mike, You might want to try putting them in different profiles (maybe one on port 5080, one on 5082?) so that the provider sees them as coming from distinct places - that way they should let you use both at once, rather than just seeing whichever was the last to register. Cheers -- Dave Here they are, mildly modified: ~/conf/sip_profiles/external/gateway1212.xml (name changed): include gateway name=33.44.55.66 param name=username value=8158381212/ param name=password value=bloodyblahbloodyblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include ~/conf/sip_profiles/external/gateway1234.xml (name changed for pseudo tn): include gateway name=33.44.55.66 param name=username value=8158381234/ param name=password value=blahbloodyblahblahblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include The gateway name is the same for both. Note that each registers and passes calls both ways on their own. I'd like to use 'em at the same time. Thanks for having a look. Mike G. On Tue, Sep 29, 2009 at 11:21 AM, Frank Carmickle fr...@carmickle.com wrote: On Tue, Sep 29, Michael Gende wrote: Say, could someone please direct me to information on registering more than one DID with a SIP provider? When I try this using two XML files in ~/conf/sip_profiles/external, I find only one or the other registers (both work fine when I use them separately). Please post your configs with out passwords. Thanks. --FC ___ 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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] Alias user mapping
On Tue, Sep 29, 2009 at 7:19 AM, Jonathan Barou jba...@sqli.com wrote: I'm sorry but I'm new in the freeswitch communauty, what I have to do to test this in svn trunk ? Thanks FYI, If you're running in Linux then this is a handy way to build from scratch using a somewhat automated process: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install What's nice about being on latest SVN trunk is that you can do make current and it will update your build for you. -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] More than One DID
Thanks for the advice, Dave. Most appreciated. Regards, Mike G. On Tue, Sep 29, 2009 at 12:11 PM, David Knell d...@3c.co.uk wrote: Hi Mike, You might want to try putting them in different profiles (maybe one on port 5080, one on 5082?) so that the provider sees them as coming from distinct places - that way they should let you use both at once, rather than just seeing whichever was the last to register. Cheers -- Dave ___ 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] More than One DID
I'll try it. Many thanks. On Tue, Sep 29, 2009 at 12:03 PM, Frank Carmickle fr...@carmickle.comwrote: On Tue, Sep 29, Michael Gende wrote: Here they are, mildly modified: ~/conf/sip_profiles/external/gateway1212.xml (name changed): include gateway name=33.44.55.66 param name=username value=8158381212/ param name=password value=bloodyblahbloodyblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include ~/conf/sip_profiles/external/gateway1234.xml (name changed for pseudo tn): include gateway name=33.44.55.66 param name=username value=8158381234/ param name=password value=blahbloodyblahblahblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include The gateway name is the same for both. Note that each registers and passes calls both ways on their own. I'd like to use 'em at the same time. This is what I was suspecting. You can't have two gateways with the same name. If you need to have them be the same domain set that with the option param name=realm value=33.44.55.66/ HTH --FC ___ 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] Freeswitch Failover
This topic has been beaten to death recently. Search the archives for things like redundancy and failover and you'll see lots of discussions. The bottom line is that your needs will dictate how much time, effort, and money you are willing to sink into this. If you want professional assistance then email consult...@freeswitch.org. If you want to do your own research then I'd say start with the ClueCon videos, specifically Day 3, Presentation #5. Here's the torrent: http://files.freeswitch.org/cluecon_2009/presentations/cluecon_2009.torrent -MC On Tue, Sep 29, 2009 at 8:16 AM, Dan White dwh...@olp.net wrote: On 29/09/09 16:37 +0530, srinivasula reddy wrote: Hi All, please let me know implementation of failover in freeswitch. I'm also interested in this topic. Obviously there are some things which are harder to failover than others, like event socket connections, or agent queue states, but what about registrations and A-leg + B-leg type calls, where the media does not flow through FreeSwitch? Are there mechanisms for 'loading' a list of registrations or call states into a FreeSwitch box? -- Dan White ___ 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] More than One DID
That did it, Frank. You're good deed for the day. Mike G. On Tue, Sep 29, 2009 at 12:20 PM, Michael Gende mge...@gendesign.comwrote: I'll try it. Many thanks. On Tue, Sep 29, 2009 at 12:03 PM, Frank Carmickle fr...@carmickle.comwrote: On Tue, Sep 29, Michael Gende wrote: Here they are, mildly modified: ~/conf/sip_profiles/external/gateway1212.xml (name changed): include gateway name=33.44.55.66 param name=username value=8158381212/ param name=password value=bloodyblahbloodyblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include ~/conf/sip_profiles/external/gateway1234.xml (name changed for pseudo tn): include gateway name=33.44.55.66 param name=username value=8158381234/ param name=password value=blahbloodyblahblahblah/ param name=expire-seconds value=60/ !--/// do not register if value=false///-- param name=register value=true/ param name=register-transport value=udp/ param name=retry-seconds value=30/ !--Use the callerid of an inbound call in the from field on outbound calls via this gateway -- param name=caller-id-in-from value=false/ param name=contact-params value=tport=5060/ !--send an options ping every x seconds, failure will unregister and/or mark it down-- param name=ping value=25/ /gateway /include The gateway name is the same for both. Note that each registers and passes calls both ways on their own. I'd like to use 'em at the same time. This is what I was suspecting. You can't have two gateways with the same name. If you need to have them be the same domain set that with the option param name=realm value=33.44.55.66/ HTH --FC ___ 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] Multiple Subnets
Hello Frank, There is only one interface on FS box, it has an IP address 192.168.50.14/24 and the 192.168.60/24 is accessible through 192.168.50.3. Correct me if I'm wrong, but if I set the mask to be /16, then 192.168.60/24 will be unreachable from FS as it will be treated at local connected network. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Frank Carmickle Sent: Tuesday, September 29, 2009 8:08 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Multiple Subnets On Tue, Sep 29, Andrey Nepomnyaschih wrote: Hello, I have some problems running a FreeSWITCH with endpoints located in different subnets. For example a FS is listening at 192.168.50.14/32 and endpoints from the same (192.168.50.0/24) subnet work as expected. But when I try to receive a media from an endpoint located at different subnet, let's say 192.168.60.0/24 the RTP stream cannot be bridged. FS sends an INVITE message asking endpoint to send media to external interface. Can anyone give me a hint where to look on how to overcome such behavior? Are we talking about different interfaces, physical or virtual? If so you must use separate profiles for each. If not then set the netmask on the interface to /16. --FC ___ 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] Searching Mailing List Archives
http://dir.gmane.org/gmane.comp.telephony.freeswitch.user also works. Many ways to find what you are after more of matter of preference and which you remember when you need to find something. -- W ___ 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] Multiple Subnets
On Tue, Sep 29, Andrey Nepomnyaschih wrote: Hello Frank, There is only one interface on FS box, it has an IP address 192.168.50.14/24 and the 192.168.60/24 is accessible through 192.168.50.3. Correct me if I'm wrong, but if I set the mask to be /16, then 192.168.60/24 will be unreachable from FS as it will be treated at local connected network. You are correct. So I guess we'll wait for Brian to look at the trace to see what your router is doing with the bits. If you have access to the router maybe having a look at it's configuration may show you what is the issue. --FC ___ 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] Problem with gateway registration
Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a Trying reply immediately followed by a 401 Unauthorized which contains a WWW-Authenticate: digest with a qop=auth parameter. Then Freeswitch replies with a second REGISTER including a large Authorization: digest section with cnonce and nc=0001 parameters. The gateway which doesn't register, doesn't send the qop=auth parameter together with the 401 Unauthorized, and then Freeswitch sends a Authorization: digest section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this qop=auth parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an Authorization: digest section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing sofia profile external restart reloadxml on the cli, captured with tshark -i eth0 -o rtp.heuristic_rtp: TRUE -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38' Thanks! Nicolas or_vs_red.pcap Description: Binary data ___ 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] Multiple Subnets
Yeah, I've been working with Network Admin on this issue, and he says that the router simply routes the packets without doing any translation stuff. Although he couldn't answer where exactly the RTP stream is going to, so we agreed to switch from one software to another tomorrow and see if makes any difference. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Frank Carmickle Sent: Tuesday, September 29, 2009 8:53 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Multiple Subnets On Tue, Sep 29, Andrey Nepomnyaschih wrote: Hello Frank, There is only one interface on FS box, it has an IP address 192.168.50.14/24 and the 192.168.60/24 is accessible through 192.168.50.3. Correct me if I'm wrong, but if I set the mask to be /16, then 192.168.60/24 will be unreachable from FS as it will be treated at local connected network. You are correct. So I guess we'll wait for Brian to look at the trace to see what your router is doing with the bits. If you have access to the router maybe having a look at it's configuration may show you what is the issue. --FC ___ 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] domain doesn't reflect the changes
hi folks, I want to change the default domain name which is by default
$${local_ip_v4} .
In the vars.xml , i have changed the value to the myexample.com but when I
restart the freeswitch server, it still shows domain to be my local ip
address.
X-PRE-PROCESS cmd=set data=domain=myexample.com/
with regards
Pankaj anand
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Re: [Freeswitch-users] Problem with gateway registration
900 level errors are sofia internal errors so probably something is wrong with your gateway config xml. if you want to send it with any critical info replaced with XXX maybe we can see the issue for you. On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner nico...@medularis.comwrote: Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a Trying reply immediately followed by a 401 Unauthorized which contains a WWW-Authenticate: digest with a qop=auth parameter. Then Freeswitch replies with a second REGISTER including a large Authorization: digest section with cnonce and nc=0001 parameters. The gateway which doesn't register, doesn't send the qop=auth parameter together with the 401 Unauthorized, and then Freeswitch sends a Authorization: digest section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this qop=auth parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an Authorization: digest section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing sofia profile external restart reloadxml on the cli, captured with tshark -i eth0 -o rtp.heuristic_rtp: TRUE -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38' Thanks! Nicolas ___ 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 msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@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
[Freeswitch-users] Siptapi and Freeswitch
Anybody tried siptapi with freeswitch? http://sourceforge.net/projects/siptapi/ This may enable Click2Dial e.g. from Outlook to Freeswitch. So anybody has experience with that solution? Best regards 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
Re: [Freeswitch-users] domain doesn't reflect the changes
On Tue, Sep 29, pankaj anand wrote:
hi folks, I want to change the default domain name which is by default
$${local_ip_v4} .
In the vars.xml , i have changed the value to the myexample.com but when I
restart the freeswitch server, it still shows domain to be my local ip
address.
I had this trouble also.
X-PRE-PROCESS cmd=set data=external_sip_ip=stun:stun.freeswitch.org/
X-PRE-PROCESS cmd=set data=external_rtp_ip=stun:stun.freeswitch.org/
were my problem. If you need stun then you will have to get your reverse dns
changed.
--FC
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Re: [Freeswitch-users] Attended transfer - no audio
Yes, the phones are behind NAT. Freeswitch is on a public IP. j On Tue, Sep 29, 2009 at 3:10 PM, Brian West br...@freeswitch.org wrote: NAT involved? /b On Sep 29, 2009, at 7:27 AM, Jan Kubr wrote: I commented out the following in our internal profile: param name=media-option value=bypass-media-after-att-xfer/ Which helped. We don't use bypass media though.. Jan ___ 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] Siptapi and Freeswitch
Nope but I would be happy to see someone try and document it on our wiki. /b On Sep 29, 2009, at 1:44 PM, Peter P GMX wrote: Anybody tried siptapi with freeswitch? http://sourceforge.net/projects/siptapi/ This may enable Click2Dial e.g. from Outlook to Freeswitch. So anybody has experience with that solution? Best regards 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
Re: [Freeswitch-users] Freeswitch Failover
I've downloaded 0Kb in 56 minutes. Anyone mind setting up a seeder? I'm on a pretty fast connection. While waiting, I found the following discussions: I found the following threads while searching for redundancy and failover: http://lists.freeswitch.org/pipermail/freeswitch-users/2007-September/001499.html (round robin) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-April/013069.html (load balancing, scalability) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/017349.html (a discussion of setting up multiple FS boxes in multiple datacenters) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-December/009723.html (comparing FS to a commercial product) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011770.html (WAN redundancy) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/010924.html (Comparing against commercial SBCs) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006710.html (Hardware redundancy) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-January/001934.html (Generic heartbeat failover question) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/017972.html (Failing over an originating call based on originate_timeout) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-August/005740.html (HA via Ultra Monkey) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-April/013097.html (gateway redundancy) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-August/005688.html (redundancy via openser) http://lists.freeswitch.org/pipermail/freeswitch-users/2007-August/001363.html (failover based on xml-curl) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-July/016904.html (multiple gateways) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/002146.html (dead gateway detection) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-June/003979.html (load balancing and failover via mod_xml_curl) http://lists.freeswitch.org/pipermail/freeswitch-users/2007-September/001445.html (multiple gateways) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-June/015363.html (hot failover) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/014071.html (failover extension) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/008532.html (an interesting discussion on failing over registrations) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-October/thread.html#7734 (clustering via DNS SRV) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/007845.html (hot failover and 6 9s!) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/005874.html (high availability clustering) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-October/007376.html (outbound call failover) Of these, the two discussing hot failover are closest to what I was referring to. Last night, we upgraded our Acme Packet Session Border Controllers. We upgraded to a new major version, including rebooting each one in turn. No registrations (and presumably no calls) were dropped during the process. They are tied together via ethernet. An SBC's function is admittedly much more narrowly focused. Has there been any work in FS in this area? I presume there is a mechanism for dumping a registration database from a FS box. Is there a way to load that database into another one? Thanks. On 29/09/09 10:34 -0700, Michael Collins wrote: This topic has been beaten to death recently. Search the archives for things like redundancy and failover and you'll see lots of discussions. The bottom line is that your needs will dictate how much time, effort, and money you are willing to sink into this. If you want professional assistance then email consult...@freeswitch.org. If you want to do your own research then I'd say start with the ClueCon videos, specifically Day 3, Presentation #5. Here's the torrent: http://files.freeswitch.org/cluecon_2009/presentations/cluecon_2009.torrent -- Dan White BTC Broadband ___ 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] Control of BLF Capabilty?
Is there a way in FS to selectively deny a BLF presence subscription request for the sake of privacy? So that groups could be defined that are allowed to be monitor or be monitored? And others that are not allowed to monitor or be monitored? 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] Control of BLF Capabilty?
Not at this time... there is nothing in place to do so. Maybe you can post a bounty on Jira for this feature. Thanks, Brian On Sep 29, 2009, at 2:52 PM, Jerry Richards wrote: Is there a way in FS to selectively deny a BLF presence subscription request for the sake of privacy? So that groups could be defined that are allowed to be monitor or be monitored? And others that are not allowed to monitor or be monitored? 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] Freeswitch Failover
Those discussions are probably it. It goes something like: I want hot failover. Can't FS just load state? No, we think it'd take about $100K of work, and considerable time Oh. XXX does it. I'd think we just need a database. No, it'll take a lot of effort. Just use a SIP proxy and redirect traffic around. Run FS on OpenVZ to avoid scheduled hardware maintenance. There's a money back guarantee on FS anyways. So, until someone gets funding and dev time scheduled, it probably won't go anywhere. Now if someone shows up with the people to actually implement it and wants some pointers on where to start after having gotten into the FS core, then the responses might be slightly different. -Michael -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Dan White Sent: Tuesday, September 29, 2009 1:05 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch Failover I've downloaded 0Kb in 56 minutes. Anyone mind setting up a seeder? I'm on a pretty fast connection. While waiting, I found the following discussions: I found the following threads while searching for redundancy and failover: http://lists.freeswitch.org/pipermail/freeswitch-users/2007-September/001499.html (round robin) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-April/013069.html (load balancing, scalability) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/017349.html (a discussion of setting up multiple FS boxes in multiple datacenters) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-December/009723.html (comparing FS to a commercial product) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011770.html (WAN redundancy) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/010924.html (Comparing against commercial SBCs) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006710.html (Hardware redundancy) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-January/001934.html (Generic heartbeat failover question) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/017972.html (Failing over an originating call based on originate_timeout) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-August/005740.html (HA via Ultra Monkey) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-April/013097.html (gateway redundancy) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-August/005688.html (redundancy via openser) http://lists.freeswitch.org/pipermail/freeswitch-users/2007-August/001363.html (failover based on xml-curl) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-July/016904.html (multiple gateways) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/002146.html (dead gateway detection) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-June/003979.html (load balancing and failover via mod_xml_curl) http://lists.freeswitch.org/pipermail/freeswitch-users/2007-September/001445.html (multiple gateways) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-June/015363.html (hot failover) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/014071.html (failover extension) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/008532.html (an interesting discussion on failing over registrations) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-October/thread.html#7734 (clustering via DNS SRV) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/007845.html (hot failover and 6 9s!) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/005874.html (high availability clustering) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-October/007376.html (outbound call failover) Of these, the two discussing hot failover are closest to what I was referring to. Last night, we upgraded our Acme Packet Session Border Controllers. We upgraded to a new major version, including rebooting each one in turn. No registrations (and presumably no calls) were dropped during the process. They are tied together via ethernet. An SBC's function is admittedly much more narrowly focused. Has there been any work in FS in this area? I presume there is a mechanism for dumping a registration database from a FS box. Is there a way to load that database into another one? Thanks. On 29/09/09 10:34 -0700, Michael Collins wrote: This topic has been beaten to death recently. Search the archives for things like redundancy and failover and you'll see lots of discussions. The bottom line is that your needs will dictate how much time, effort, and money you are willing to sink into this. If you want professional assistance then email consult...@freeswitch.org. If you want to do your own research then I'd say start with the
[Freeswitch-users] Outbound INVITE rejected with 480 Temp Unavail, Reason MANDATORY_IE_MISSING
Hello All, I have an internal extension that needs to send an INVITE without SDP body (Content Length 0). Freeswitch is replying with 480 Temporarily Unavailable with reason MANDATORY_IE_MISSING. Would anyone know what I need to do to enable this? 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] Multitenancy
This is how I went and made the multi-tenant, this config will simply create two users: 1...@company-a.org and 1...@company-b.org. It also has two dialplan/context for each domain, so if the 1...@company-a.org user tries to dial a extension, it will go and look in the company-a.org context/dialplan, etc. http://pastebin.freeswitch.org/10513 http://pastie.org/635706 Hope that helps someone. Diego On Wed, Sep 23, 2009 at 8:22 PM, Diego Viola diego.vi...@gmail.com wrote: Ok, sorry for that and thanks for the help :). Diego On Wed, Sep 23, 2009 at 8:09 PM, Brian West br...@freeswitch.org wrote: Can you next time pause a few moments... Think about what you're sending and send ONE email with your questions? This 10 emails from you replying to yourself things looks like you're a crazy man! :P /b PS: ask on IRC or mailing list NOT BOTH please. On Sep 23, 2009, at 2:48 PM, Diego Viola wrote: I prefer to specify the context as a per-domain so it affects all the users on the domain directly... On Wed, Sep 23, 2009 at 7:48 PM, Diego Viola diego.vi...@gmail.comwrote: Do I specific the context as a per-user thing, can I specific the context as a per-domain way? Diego On Wed, Sep 23, 2009 at 7:42 PM, Diego Viola diego.vi...@gmail.comwrote: s/directory/directories/ Should I use context for that? On Wed, Sep 23, 2009 at 7:40 PM, Diego Viola diego.vi...@gmail.comwrote: Ok I have configured the two domains with their own directory and I can register fine with them now. But I need to configure two different dialplans with their own profiles. How do I tell a specific domain to use a specific profile/dialplan? Thanks, Diego On Wed, Sep 23, 2009 at 6:52 PM, Diego Viola diego.vi...@gmail.comwrote: I was having some issues with DNS, I tried to register with the new directory and domain but I got can't find user until I commented force-register-domain and force-register-db-domain from the profile. Thanks for the tip Brian :). Diego On Wed, Sep 23, 2009 at 6:16 PM, Brian West br...@freeswitch.orgwrote: Thats up to you :P /b On Sep 23, 2009, at 1:10 PM, Diego Viola wrote: Should I delete the directory default and default.xml when I copy default to foo.org and bar.org etc? Diego ___ 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 ___ 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] Multitenancy
Why not start a wiki page on the topic? /b On Sep 29, 2009, at 4:33 PM, Diego Viola wrote: Hope that helps someone. Diego ___ 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] Multitenancy
Someone already wikifyed it for me =D http://wiki.freeswitch.org/wiki/Multiple_Companies Thanks Brian :) On Tue, Sep 29, 2009 at 9:49 PM, Brian West br...@freeswitch.org wrote: Why not start a wiki page on the topic? /b On Sep 29, 2009, at 4:33 PM, Diego Viola wrote: Hope that helps someone. Diego ___ 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 gateway registration
Anthony, thanks. Below are my config files for the two gateways from the sip trace. Both files are located in conf/directory/default. - redvoiss.xml (the one that works) include user id=gateway_redvoiss gateways gateway name=redvoiss-pp param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=pxextmy.redvoiss.net/ param name=realm value=pxextmy.redvoiss.net/ param name=proxy value=pxextmy.redvoiss.net/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2010/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - orange.xml (the one that doesn't work) include user id=gateway_orange gateways gateway name=orange param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=216.72.10.39/ param name=realm value=216.72.10.39/ param name=proxy value=216.72.10.39/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2011/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - If I remove the register=true param for the non-working gateway, I don't get the registration error on the cli, but then all call attempts get rejected with a 401 Unauthorized, and I get a hangup cause of NORMAL_UNSPECIFIED. Best, Nicolas On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale anthony.miness...@gmail.com wrote: 900 level errors are sofia internal errors so probably something is wrong with your gateway config xml. if you want to send it with any critical info replaced with XXX maybe we can see the issue for you. On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner nico...@medularis.comwrote: Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a Trying reply immediately followed by a 401 Unauthorized which contains a WWW-Authenticate: digest with a qop=auth parameter. Then Freeswitch replies with a second REGISTER including a large Authorization: digest section with cnonce and nc=0001 parameters. The gateway which doesn't register, doesn't send the qop=auth parameter together with the 401 Unauthorized, and then Freeswitch sends a Authorization: digest section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this qop=auth parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an Authorization: digest section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing sofia profile external restart reloadxml on the cli, captured with tshark -i eth0 -o rtp.heuristic_rtp: TRUE -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38' Thanks! Nicolas ___ 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 msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference
Re: [Freeswitch-users] Alias user mapping
I've noticed that when using an alphanumeric ID, case sensitivity causes
issues. It doesn't seem to matter what case the userID is in the XML
directory. What does seem to matter is how the user registered their
phone. If the user registered in all upper case (JOHN_SMITH), then
dialing John_Smith will not work.
My work around for this is to tell all users to register their phones in
lower case, and then my dial plan uses LUA to force all dialed
extensions to lowercase.
Peter
Jonathan Barou wrote:
Hi everybody,
I have a problem with Alphanumeric to numeric user mapping
I have done like it's written here :
Alphanumeric to numeric user mapping
Say you want a user's id to be alphanumeric (like an email username),
such as johnsm...@pbx.example.com mailto:johnsm...@pbx.example.com.
These users have alphanumeric usernames in their sip phone config, but
you want to map them from their sip username to a numeric extension,
and vice versa.
As of version 1.0.4, Freeswitch makes this trivial to accomplish. A
user's ID can be any alphanumeric string, and this can be simply tied
to an extension number using the 'number-alias' property. This
property creates an aliased directory entry that points to the
alphanumeric user entry.
/NOTE: When using this attribute, you must be careful not to create a
directory collision by having another user whose ID is the same as
another user's alias/
Here is an example from the user directory:
user id=johnsmith number-alias=1001
!-- Insert the usual user configuration variables and params here,
including user password, voicemail password, caller ID info, etc --
/user
So when a user dials extension number 1001, your dialplan can use the
'user_data' function to look up the ID attribute associated with that
number alias. In the default dialplan, the 'Local Extension' section
can be made to work with a small change to the 'bridge' line:
action application=bridge
data=user/${user_data(${dialed_extensi...@${domain_name} attr id)}...@${domain_name}/
/NOTE: Using this user_data function in combination with mod_xml_curl
will generate an additional request each time the user_data function
is called. Note that it is already called once in the Local Extension
section to determine the callgroup. *Beware of performance
implications of this with high-volume systems.*/
But when I want to call my alias-number, FS says No Route, Abording
My version of FreeSWITCH is the 1.0.4pre9.
Do you have any ideas ? Did I forget something to do?
Thanks
--
Jonathan BAROU
SQLI LYON - CRCI
jba...@sqli.com mailto:jba...@sqli.com
lyon.c...@sqli.com mailto:lyon.c...@sqli.com
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[Freeswitch-users] REGISTER fails with 407 after minutes of success register
Hi all, I have a FS that registers on an Ericsson pabx as gateway under sip_external. This gateway start registering on the Ericsson ok, but after a while, around 50mins, it fails with the logs below. If I hit *sofia profile external restart* on fs_cli then the gateway returns to register with success (that means, we get 200 OK from Ericsson). This happens with FS 1.0.4 release tarball, and trunk r15011. I found similar situations on these links, but not actually found a solution. Any help is very welcome. *OS* CentOS 5.3 x86_64 4Gb RAM log at http://pastebin.freeswitch.org/10517 conf/sip_profiles/external/ericsson.xml: gateway name=ericsson_1064 param name=username value=1064/ param name=realm value=10.227.0.3/ param name=password value=5080/ param name=register value=true/ param name=register-proxy value=10.227.0.3/ param name=expire-seconds value=60/ param name=retry-seconds value=10/ /gateway ___ 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] REGISTER fails with 407 after minutes of success register
I need the sip trace. /b On Sep 29, 2009, at 7:28 PM, Fernando Testa wrote: Hi all, I have a FS that registers on an Ericsson pabx as gateway under sip_external. This gateway start registering on the Ericsson ok, but after a while, around 50mins, it fails with the logs below. If I hit *sofia profile external restart* on fs_cli then the gateway returns to register with success (that means, we get 200 OK from Ericsson). This happens with FS 1.0.4 release tarball, and trunk r15011. I found similar situations on these links, but not actually found a solution. Any help is very welcome. *OS* CentOS 5.3 x86_64 4Gb RAM log at http://pastebin.freeswitch.org/10517 conf/sip_profiles/external/ericsson.xml: gateway name=ericsson_1064 param name=username value=1064/ param name=realm value=10.227.0.3/ param name=password value=5080/ param name=register value=true/ param name=register-proxy value=10.227.0.3/ param name=expire-seconds value=60/ param name=retry-seconds value=10/ /gateway ___ 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] Need to know
Hi, I want to use mod_nibble in freeswitch for billing purpose. I've got some questions, as I'm listing down below; 1- How I create MySQL connection with FS? 2- Is it possible to query the database when billing occurs? 3- Will database credentials require each time when query to database? 4- I'm currently running MySQL database in production for billing purposes.Will it be okay to use my current database for mod_nibble? Kindly advise me soon and also please provide me the links as well. -- Regards, Ahmed Munir ___ 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] Need to know
mod_nibble uses ODBC. Via ODBC u can connect to mysql. http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core mod_nibble has it's own table design. http://wiki.freeswitch.org/wiki/Mod_nibblebill It may be possible to tweak it to use your current tables but I'd suggest getting the basics working based on the documentation on the wiki and build from there. -- W ___ 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] Put a call in music on hold
Dear all, I am in the process of implementing IVR server in Perl using event outbound socket. I want to put a call in music on hold when the perl is doing something. I have tried the following steps. * First using Set application set the hold_music variable with our music file * Call the uuid_hold application with current call unique id. * Wait 10 seconds * Call the uuid_hold application with off and current call unique id I executed the server and called up the configured extension. It didn't work. I also tried with nc command. And send the events thorough the terminal. At that time also it didn't work. Did I make any mistake? Is it possible to put a call on hold and play the voice while holding time? Please help me? -- Regards, Thangappan.M ___ 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
