Re: [Freeswitch-users] SIP dump to DB
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, maybe this helps: http://www.wesip.com/mediawiki/index.php/SipSpy regards helmut On 17.05.2009 20:33, Ron McCarthy wrote: Kokoska, Did you ever find a solution for this? I have been working on this as well, trying to write some perl application to read the data from ngrep and parse it, but have got no where. I hope you have better luck then I have! On Mon, Feb 23, 2009 at 11:13 PM, kokoska.rokoska kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote: Joseph Bajin napsal(a): If you write it correctly it will work just fine. Yes, this is challenge I have talked about :-) That is how most of all the other correlation engines work. I don't have enough informations but from what I heard from friendly competitors they are usualy log (SIP|ISUP) messages after they are parsed by their routing servers and not run separate tshark+parser+logger. Or they duplicate (just) SIP messages to separate machine and parse and log them there (SERlike server + sip_trace). Your setup is not going to be bigger than some of the large telecoms that use these systems today. I hope so :-) Thanks once more, Joseph, for your info! Best regards, kokoska.rokoska ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto: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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKEQ9m4tZeNddg3dwRAofUAJ4nRE8237OQQyb2ybzDiAyH4XEYIgCffmry QDyWbwRq/0IHQTRW+i/yhH4= =w3UC -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] Unable to successfully bridge calls to an external user
Ok, my FS box is on a LAN IP (10.0.0.12), but it is setup as the DMZ
machine on my modem so it can receive incoming connections without any
NAT related problems.
I'm trying to get a user outside on the internet to connect to my FS
box and register as an internal user. He is using X-Lite on his laptop
behind his own NAT. His external IP is 203.206.171.118.
His registration looks like this:
Call-ID:NGU1NjE4ZmIxODMwMWM1OTZlMDYzMzk0NjMyMjM1YWY.
User: 1...@10.0.0.12
Contact:124.254.81.250
sip:1...@203.206.171.118:40168;rinstance=c5779e159bbe8bc7
Agent: X-Lite release 1014k stamp 47051
Status: Registered(UDP)(unknown) EXP(2009-05-18 19:32:03)
Host: kira
IP: 203.206.171.118
Port: 40168
Auth-User: 1001
Auth-Realm: 124.254.81.250
I note that it's registered as plain UDP, not UDP-NAT like my own
internal extensions are.
The dialplan is set to route this DID (0746029001) to user 1...@$$
{domain} as follows:
extension name=Jake
condition field=destination_number
expression=^(0746029001)$
action application=bridge
data=USER/1...@$${domain}/
/condition
/extension
When I try and make a call from my mobile (0451282630) to the DID, it
says it's bridging to USER/1...@10.0.0.12, but when the person
answers, we get no audio in either direction. It rings and answers
fine, it just doesn't send any audio in either direction so I'm
suspecting a bridging problem.
The log file of the connection is on the web at
http://pastebin.freeswitch.org/8990
The bridge line is:
EXECUTE sofia/external/0451282...@203.161.130.132 bridge(USER/1...@10.0.0.12
)
But the sofia address for the connection is shown as
sofia/internal/sip:1...@203.206.171.118:40168;rinstance=c5779e159bbe8bc7
Is this correct ? Am I missing something fundamental ? His user
address is @10.0.0.12, but his sofia address is sip:
1...@203.206.171.118. Is this OK or should his user ID be at his
actual ip address ? This seems normal to me as I believe the 10.0.0.12
address is the domain of the FS box. Is it OK that he's in the same
domain as my own users on my LAN or am I supposed to configure a
different domain for him because he's outside.
I thought maybe it was a double-NAT problem, but the log doesn't show
any fs_nat=yes entries so I assume it's not trying to NAT him (as it
shouldn't). The situation is an external mobile rings my DID, so the
call comes in from my provider's address, hits my FS box, which
successfully sends at least the ringing information out to his
softphone at his external IP, but then when it bridges, it seems not
to send the audio to the right place.
I'm terrible with FS log files so I have no idea whether any of the
entries are wrong. What's likely to be my issue here ? Is it NAT-
related, or routing related ? Any suggestions appreciated.
David
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[Freeswitch-users] openzap and progress detection
Hello, I'm in Spain with an analog TDM400 Clone from OpenVox (with 1xFXO +1xFXS), and am trying first to make the FXO work with Openzap and Freeswitch (using dahdi 2.2.0-rc4). Openzap perfectly detects and loads the spans, but I'm currently enable to dial out with the FXO module, it doesn't dial anything and times-out after 30 seconds. I believe it has to do with some tone detection and therefore have a few questions: - When I plug in the line, dahdi sends an event (event 17) that is ignored by OZ. Can we enable some type of battery check in OZ before dialing out, or is there some variable to monitor battery (oz dump doesn't show the battery status)? - Does OZ use the polarity switch events sent by dahdi (in kewlstart mode) for answer and hangup detection? - Apparently, OZ does tone progress detection by frequency, but here in Spain, most tones use the same 425Hz frequency with different on-off timing. Is it possible to detect those? - As some PBX in Spain transfer calls by first hanging up and picking up on another phone, can we enable/disable parts of polarity switch and/or tone progress detection (ex: (hangup)/(answer)onpolarityswitch in Asterisk) - Some lines here are connected to very old FXS from operators that have low sound quality and can take a few seconds to give a dial tone when picking up. Is it possible to introduce a delay before sending DTMF digits when dialing? Is it possible to relax DTMF detection, and tweak DTMF settings (make them a bit longer, with a longer pause for the other side to detect)? My ideal case to make it work in every case around here would be to: - have OZ fail to dial if battery is not present (and be able to fetch battery status somehow) - disable tone progress (sometimes call ends up on some local PBX that answers and provides US tones which are different) - be able to have an initial pause before dialing with DTMF digits - use polarity switch to detect remote answer, but not hangup (for transfer issues) Is the above possible? Thanks in advance, François. ___ 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] Unable to successfully bridge calls to an external user
David Robinson pawzl...@gmail.com wrote: Is this correct ? Am I missing something fundamental ? My suspicion is that the RTP traffic isn't traversing the NAT properly. You may have to configure the routers at both ends to forward the RTP packets to the correct destinations. There is a good discussion of NAT on the wiki. One of the great advantages of IPv6 is that NAT goes away altogether. I've been achieving quite reasonable call quality even across IPv6-over-IPv4 tunnels. ___ 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 Drops Call after 30 Seconds
Hi Adam, We had exactly the same problem which we initially believed to be a NAT firewall issue. However, I changed my firewall to transparent mode and the problem still persisted. In the end, I solved the problem by changing VOIP provider. I was using AQL which I couldn't make work and now use VOIPTALK. Cheers Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of adamF Sent: 17 May 2009 19:12 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch Drops Call after 30 Seconds I am having an issue where freeswitch drops an incoming call after being connected for 30 seconds. Here is my console output starting from the termination sequence. Any help would be greatly appreciated. 2009-05-17 10:59:41 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/external/2062842...@64.24.35.74 entering state [terminating] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:355 audio_bridge_thread() sofia/external/2062842...@64.24.35.74 ending bridge by request from write function 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:410 audio_bridge_thread() sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes receive message [UNBRIDGE] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/external/2062842...@64.24.35.74 [BREAK] 2009-05-17 10:59:41 [NOTICE] switch_ivr_bridge.c:471 audio_bridge_on_exchange_media() Hangup sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-05-17 10:59:41 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [KILL] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:467 switch_core_session_run() (sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) State EXCHANGE_MEDIA going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) Running State Change CS_HANGUP 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) State HANGUP 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes hanging up, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE to sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes Standard HANGUP, cause: NORMAL_CLEARING 2009-05-17 10:59:41 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) State HANGUP going to sleep 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 4 (sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_na t=yes) Locked, Waiting on external entities 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:410 audio_bridge_thread() sofia/external/2062842...@64.24.35.74 receive message [UNBRIDGE] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/external/2062842...@64.24.35.74 [BREAK] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:436 audio_bridge_thread() BRIDGE THREAD DONE [sofia/external/2062842...@64.24.35.74] 2009-05-17 10:59:41 [DEBUG] switch_ivr_bridge.c:440 audio_bridge_thread() Send signal sofia/internal/sip:1...@192.168.1.10:50130;rinstance=4ddf03d1b8fc5c02;fs_nat =yes [BREAK] 2009-05-17 10:59:41 [NOTICE] switch_ivr_bridge.c:960 switch_ivr_multi_threaded_bridge() Hangup sofia/external/2062842...@64.24.35.74 [CS_EXECUTE] [NORMAL_CLEARING] 2009-05-17 10:59:41 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/external/2062842...@64.24.35.74 [KILL] 2009-05-17 10:59:41 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal
Re: [Freeswitch-users] Unable to successfully bridge calls to an external user
Can you post the INVITE and 200 OK messages from your mates end of the
call. Even if you forward the ports on the router, the RTP will not
traverse correctly if the advertised IP address is an internal one for
both ends.
On Mon, May 18, 2009 at 6:20 PM, David Robinson pawzl...@gmail.com wrote:
Ok, my FS box is on a LAN IP (10.0.0.12), but it is setup as the DMZ
machine on my modem so it can receive incoming connections without any
NAT related problems.
I'm trying to get a user outside on the internet to connect to my FS
box and register as an internal user. He is using X-Lite on his laptop
behind his own NAT. His external IP is 203.206.171.118.
His registration looks like this:
Call-ID: NGU1NjE4ZmIxODMwMWM1OTZlMDYzMzk0NjMyMjM1YWY.
User: 1...@10.0.0.12
Contact: 124.254.81.250
sip:1...@203.206.171.118:40168;rinstance=c5779e159bbe8bc7
Agent: X-Lite release 1014k stamp 47051
Status: Registered(UDP)(unknown) EXP(2009-05-18 19:32:03)
Host: kira
IP: 203.206.171.118
Port: 40168
Auth-User: 1001
Auth-Realm: 124.254.81.250
I note that it's registered as plain UDP, not UDP-NAT like my own
internal extensions are.
The dialplan is set to route this DID (0746029001) to user 1...@$$
{domain} as follows:
extension name=Jake
condition field=destination_number
expression=^(0746029001)$
action application=bridge
data=USER/1...@$${domain}/
/condition
/extension
When I try and make a call from my mobile (0451282630) to the DID, it
says it's bridging to USER/1...@10.0.0.12, but when the person
answers, we get no audio in either direction. It rings and answers
fine, it just doesn't send any audio in either direction so I'm
suspecting a bridging problem.
The log file of the connection is on the web at
http://pastebin.freeswitch.org/8990
The bridge line is:
EXECUTE sofia/external/0451282...@203.161.130.132 bridge(USER/1...@10.0.0.12
)
But the sofia address for the connection is shown as
sofia/internal/sip:1...@203.206.171.118:40168;rinstance=c5779e159bbe8bc7
Is this correct ? Am I missing something fundamental ? His user
address is @10.0.0.12, but his sofia address is sip:
1...@203.206.171.118. Is this OK or should his user ID be at his
actual ip address ? This seems normal to me as I believe the 10.0.0.12
address is the domain of the FS box. Is it OK that he's in the same
domain as my own users on my LAN or am I supposed to configure a
different domain for him because he's outside.
I thought maybe it was a double-NAT problem, but the log doesn't show
any fs_nat=yes entries so I assume it's not trying to NAT him (as it
shouldn't). The situation is an external mobile rings my DID, so the
call comes in from my provider's address, hits my FS box, which
successfully sends at least the ringing information out to his
softphone at his external IP, but then when it bridges, it seems not
to send the audio to the right place.
I'm terrible with FS log files so I have no idea whether any of the
entries are wrong. What's likely to be my issue here ? Is it NAT-
related, or routing related ? Any suggestions appreciated.
David
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[Freeswitch-users] ODBC and Core-DB
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, does anybody know if and how FS can export its core db to an external database via odbc like mod_limit or mod_sofia? If not, is such a feature planned for the near future? regards Helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKEVeI4tZeNddg3dwRAuMEAJ0dWJYGNDVDdtqJJuBRU/MFAtfAYQCeMqkR 7jEHwH2bxWEb/ccbdajxt5U= =XYMu -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
Re: [Freeswitch-users] ODBC and Core-DB
No, you can't. Math On 18-May-09, at 2:41 PM, Helmut Kuper wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, does anybody know if and how FS can export its core db to an external database via odbc like mod_limit or mod_sofia? If not, is such a feature planned for the near future? regards Helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKEVeI4tZeNddg3dwRAuMEAJ0dWJYGNDVDdtqJJuBRU/MFAtfAYQCeMqkR 7jEHwH2bxWEb/ccbdajxt5U= =XYMu -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] Logging 503's or other errors
enable the b leg cdr as well and you will also get cdr from the b leg perspective. both xml cdr and cdr csv have params in the config to enable it. On Sun, May 17, 2009 at 1:26 PM, Ron McCarthy ronmc...@gmail.com wrote: Hi list, Ive been trying to find a way to log 503's, 480's and other SIP response codes. If we have continue_on_fail=true and have multiple gateways for the call to go out, if the 1st,2nd or whatever gateways fail can we log it somehow? We'd like to know if a carrier is having issues or not letting us send calls for some reason, from what I can tell I only show one CDR get written and that's at the end of the call, so it says nothing about the gateways we tried to send a call before and if they failed. Any ideals on how to do this? Im using the XML CURL dialplan if that matter. Any ideals how this could be setup so we can keep track of what is going on? 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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] freeswitch and invites without sdp in initial invite
Hi there, Freeswitch looks really interesting. I am trying to connect a softswitch wich does some strange things. If a call arrives from pots the softswitch won't send sdp information in the initial invite. Is this something I can change in the sip-profile with freeswitch? BR Uwe ___ 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 and invites without sdp in initial invite
Hi there, Freeswitch looks really interesting. I am trying to connect a softswitch wich does some strange things. If a call arrives from pots the softswitch won't send sdp information in the initial invite. Is this something I can change in the sip-profile with freeswitch? BR Uwe ___ 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] silly (?) questions
Hi list, I just discovered FS (practiced a bit * 2 years ago, but too much unstable) and find it cool, NOT CPU greedy and (almost) working ouf of the web. I'd like to know if star codes (such as *98) are normalized or not? (and if so, where I could find a list) Also, as I don't use very much my phone and mostly don't pay for it (I live in france and got unlimited free call for 70-80 countries, and my phone is actually plugged in my ADSL box) I'd like to leave access for other people through something like DUNDi (that I don't really know.) BUT not everything is free (i.e.: cellular phones calls cost €0.22 @ connection + €0.22/min); thus I must forbid this kind of calls. Does anybody have realized that, because I need a good template? Thanks JY -- Old mail has arrived. ___ 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] fixed Re: freeswitch and invites without sdp in initial invite
Hi, allowing 3cc fixes the problem. BR Uwe Uwe Kastens schrieb: Hi there, Freeswitch looks really interesting. I am trying to connect a softswitch wich does some strange things. If a call arrives from pots the softswitch won't send sdp information in the initial invite. Is this something I can change in the sip-profile with freeswitch? BR Uwe ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ 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] silly (?) questions
Hello Jean-Yves,
did you ever try a call-trough? (a person dials in (1234567, see below)
types the target number as DTMF and gets connected to this number?
A basic dialplan can be like this:
extension name=Dialthru
condition field=destination_number expression=^(1234567)$
action application=play_and_get_digits data=5 25 3 7000 #
ivr/8000/ivr-enter_ext.wav voicemail/8000/vm-that_was_an_invalid_ext.wav
foobar \d+/
action application=transfer data=${foobar} XML default/
/condition
/extension
For \d+ you may define your regular expression, which numbers you
would accept. Also you may try to redirect into the dialplan again after
the number is entered (instead of directly transferring the call).
Best regards
Peter
Jean-Yves F. Barbier schrieb:
Hi list,
I just discovered FS (practiced a bit * 2 years ago, but too much unstable)
and find it cool, NOT CPU greedy and (almost) working ouf of the web.
I'd like to know if star codes (such as *98) are normalized or not?
(and if so, where I could find a list)
Also, as I don't use very much my phone and mostly don't pay for it (I
live in france and got unlimited free call for 70-80 countries, and
my phone is actually plugged in my ADSL box) I'd like to leave access
for other people through something like DUNDi (that I don't really
know.)
BUT not everything is free (i.e.: cellular phones calls cost €0.22 @
connection + €0.22/min); thus I must forbid this kind of calls.
Does anybody have realized that, because I need a good template?
Thanks
JY
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[Freeswitch-users] crash-protection and monit
Hi, Is there any reason why the crash-protection parameter in switch.conf.xml defaults to false and are there any downsides to setting it to true? The documentation says it helps with certain types of crashes, can anyone tell me what sort of crashes in particular it helps to prevent as my freeswitch install seems to crash every few days. Also, does anyone have an example of the monit setup for freeswitch to restart it when it fails? Many thanks Andy ___ 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] crash-protection and monit
Hi, Crash protection catches segmentation faul signals and try to kill that thread only. It works for stupid errors like a null pointer dereference, but in most scenarios, a crash means something in the process memory was corrupted. Ignoring it will just make it crash later on, thats why the default is false. Now if you have a crash, you should update to svn trunk and if it still happens, report it on Jira ( see http://wiki.freeswitch.org/wiki/Reporting_Bugs ) so someone can look at it and fix it. Math On 18-May-09, at 6:21 PM, Andy Ayers wrote: Hi, Is there any reason why the crash-protection parameter in switch.conf.xml defaults to false and are there any downsides to setting it to true? The documentation says it helps with certain types of crashes, can anyone tell me what sort of crashes in particular it helps to prevent as my freeswitch install seems to crash every few days. Also, does anyone have an example of the monit setup for freeswitch to restart it when it fails? Many thanks Andy ___ 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] User Directory and Per-user (Channel) variables
bridge() appears to be ignoring the absolute_codec_string channel variable
defined in the User Directory even though info shows that it is present.
Other variables, such as effective_caller_id_number seem to behave
correctly which leads me to believe that this may be a very minor bug.
In order to ease trouble shooting, I have tried to implement it using a
configuration that clings rather closely to the sample/default configuration
files...
// User Directory sample
user id=5551212 mailbox=5551212
params
param name=password value=5551212/
param name=vm-password value=5551212/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=user_context value=default/
variable name=effective_caller_id_name value=5551212/
variable name=effective_caller_id_number value=5551212/
variable name=absolute_codec_string value=PCMU/
/variables
/user
// Dialplan broken sample (relies on the channel variable defined in the
User Directory)
extension name=domestic.example.com
condition field=${toll_allow} expression=domestic/
condition field=destination_number expression=^(\d{11})$
action application=info/
action application=bridge
data=sofia/gateway/${default_gateway}/$1/
/condition
/extension
// Dialplan working sample (explicit use of the channel variable)
extension name=domestic.example.com
condition field=${toll_allow} expression=domestic/
condition field=destination_number expression=^(\d{11})$
action application=info/
action application=set
data=effective_caller_id_number=${outbound_caller_id_number}/
action application=set
data=effective_caller_id_name=${outbound_caller_id_name}/
action application=bridge
data={absolute_codec_string=PCMU}sofia/gateway/${default_gateway}/$1/
/condition
/extension
// Gateway sample
gateway name=pstn
param name=username value=outbound/
param name=password value=outbound/
param name=proxy value=192.168.1.1/
param name=register value=false/
param name=caller-id-in-from value=true/
/gateway
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Re: [Freeswitch-users] User Directory and Per-user (Channel) variables
Because by the time it gets here... the codec is already picked.. you'll have to turn on late neg. for this to work. /b On May 18, 2009, at 1:08 PM, Metik wrote: variable name=absolute_codec_string value=PCMU/ Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] User Directory and Per-user (Channel)variables
Oddly enough, I initially though that was the problem and enabled it without any success... freeswi...@noesis.metik.com sofia status profile internal API CALL [sofia(status profile internal)] output: = Nameinternal Domain Name N/A DBName sofia_reg_internal Pres Hosts DialplanXML Context public Challenge Realm auto_from RTP-IP 192.168.1.100 Ext-RTP-IP 192.168.1.100 SIP-IP 192.168.1.100 Ext-SIP-IP 192.168.1.100 URL sip:mod_so...@192.168.1.100:5062 BIND-URLsip:mod_so...@192.168.1.100:5062;maddr=192.168.1.100 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGtrue PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN5 FAILED-CALLS-IN 0 CALLS-OUT 10 FAILED-CALLS-OUT0 ___ 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] Logging 503's or other errors
Even the b leg cdr is enabled it only remember the final state(channel
vars) on the b leg.
At least there are two possible ways to keep tracking all the gateways:
1) don't use '|' separated dial string, use a lua script like this:
session:execute(bridge, dial_string1);
bridge_hangup_cause =
session:getVariable(bridge_hangup_cause) or
session:getVariable(originate_disposition);
if (bridge_hangup_cause == NORMAL_TEMPORARY_FAILURE or
bridge_hangup_cause == NO_ROUTE_DESTINATION or bridge_hangup_cause
== CALL_REJECTED) then
freeswitch.consoleLog(notice, Hangup. Cause: [ ..
bridge_hangup_cause .. ]. Retry:
-- database.insert('something')
session:execue(bridge, dial_string2);
if (bridge_hangup_cause == NORMAL_TEMPORARY_FAILURE or
bridge_hangup_cause == NO_ROUTE_DESTINATION or bridge_hangup_cause
== CALL_REJECTED) then
session:execute(bridge, dial_string3);
obviously it can be done in a loop
2) by sip: add a custom header to INVITE,
bridge({sip_h_x_xxx=yyy}sofia/gateways/a/...|sofia/gateways/b/...|
sofia/gateways/c/...
be sure to give yyy a unique value each time you call, then you can
dump all the sip messages and by cross reference of the sip_h_x_xxx
and call-ID you can get all the related sip messages(every INVITE will
have the same sip_h_x_xxx header and each INVITE related message will
have the same call-ID.
On May 18, 2009, at 9:05 PM, Anthony Minessale wrote:
enable the b leg cdr as well and you will also get cdr from the b
leg perspective.
both xml cdr and cdr csv have params in the config to enable it.
On Sun, May 17, 2009 at 1:26 PM, Ron McCarthy ronmc...@gmail.com
wrote:
Hi list,
Ive been trying to find a way to log 503's, 480's and other SIP
response codes. If we have continue_on_fail=true and have multiple
gateways for the call to go out, if the 1st,2nd or whatever gateways
fail can we log it somehow? We'd like to know if a carrier is having
issues or not letting us send calls for some reason, from what I can
tell I only show one CDR get written and that's at the end of the
call, so it says nothing about the gateways we tried to send a call
before and if they failed.
Any ideals on how to do this? Im using the XML CURL dialplan if that
matter. Any ideals how this could be setup so we can keep track of
what is going on?
Thanks
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Re: [Freeswitch-users] User Directory and Per-user (Channel)variables
absolute_codec_string needs to be available from the B-leg too so it can be used on outbound channels. Add that to your directory entry and it should work. variable name=export_vars value=absolute_codec_string / Math On 18-May-09, at 9:18 PM, Brian West wrote: Are you authenticating phone calls? Also hop on IRC this email ping pong is too slow. /b On May 18, 2009, at 2:07 PM, Metik wrote: Oddly enough, I initially though that was the problem and enabled it without any success... Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] User Directory and Per-user (Channel)variables
you for sure need late negotiation:
also:
You are only setting the variable on the inbound leg but not the outbound
leg.
Remember there are 2 separate channels here.
Try this in the same place you are setting the caller id in your broken
example:
!-- this will allow absolute_codec_string to be propagated across
the downside here is that it will clobber any other previous exported
vars
-
action application=set data=export_vars=absolute_codec_string/
or
!-- This is a bit redundant but will ensure it's set on inbound and
outbound legs --
action application=export
data=absolute_codec_string=${absolute_codec_string}/
you could also do this to pass it across
action application=bridge
data={absolute_codec_string=${absolute_codec_string}}sofia/gateway/${default_gateway}/$1/
On Mon, May 18, 2009 at 2:07 PM, Metik freeswitch-users-l...@metik.comwrote:
Oddly enough, I initially though that was the problem and enabled
it without any success...
freeswi...@noesis.metik.com sofia status profile internal
API CALL [sofia(status profile internal)] output:
=
Nameinternal
Domain Name N/A
DBName sofia_reg_internal
Pres Hosts
DialplanXML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.100
Ext-RTP-IP 192.168.1.100
SIP-IP 192.168.1.100
Ext-SIP-IP 192.168.1.100
URL sip:mod_so...@192.168.1.100:5062
BIND-URLsip:mod_so...@192.168.1.100:5062
;maddr=192.168.1.100
HOLD-MUSIC local_stream://moh
OUTBOUND-PROXY N/A
CODECS G722,PCMU,PCMA,GSM
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEGtrue
PROXY-MEDIA false
AGGRESSIVENAT false
STUN-ENABLEDtrue
STUN-AUTO-DISABLE false
CALLS-IN5
FAILED-CALLS-IN 0
CALLS-OUT 10
FAILED-CALLS-OUT0
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[Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK
On register, sometimes my voip client got SIP/2.0 482 Request merged sometimes got 200 ok. 482 also means loop detected. my client only has one account logged in only one place, and no proxy, can I take 482 as 200 OK? Thanks. from RFC 3261: 8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction. ___ 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] User Directory and Per-user (Channel)variables
=D That's another way I didn't mention. There are 2 more but they are more complicated so I will omit them ;) On Mon, May 18, 2009 at 2:25 PM, Mathieu Rene mrene_li...@avgs.ca wrote: absolute_codec_string needs to be available from the B-leg too so it can be used on outbound channels. Add that to your directory entry and it should work. variable name=export_vars value=absolute_codec_string / Math On 18-May-09, at 9:18 PM, Brian West wrote: Are you authenticating phone calls? Also hop on IRC this email ping pong is too slow. /b On May 18, 2009, at 2:07 PM, Metik wrote: Oddly enough, I initially though that was the problem and enabled it without any success... Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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
Re: [Freeswitch-users] SIP/2.0 482 Request merged vs 200 OK
Is this in regards to FreeSWITCH or something else you're writing? /b On May 18, 2009, at 2:34 PM, dujinfang wrote: On register, sometimes my voip client got SIP/2.0 482 Request merged sometimes got 200 ok. 482 also means loop detected. my client only has one account logged in only one place, and no proxy, can I take 482 as 200 OK? Thanks. from RFC 3261: 8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] User Directory and Per-user(Channel)variables
Math, That was it--thank you very much! -Metik - Original Message - From: Mathieu Rene To: freeswitch-users@lists.freeswitch.org Sent: Monday, May 18, 2009 3:25 PM Subject: Re: [Freeswitch-users] User Directory and Per-user(Channel)variables absolute_codec_string needs to be available from the B-leg too so it can be used on outbound channels. Add that to your directory entry and it should work. variable name=export_vars value=absolute_codec_string / Math On 18-May-09, at 9:18 PM, Brian West wrote: Are you authenticating phone calls? Also hop on IRC this email ping pong is too slow. /b On May 18, 2009, at 2:07 PM, Metik wrote: Oddly enough, I initially though that was the problem and enabled it without any success... Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 ___ 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/2.0 482 Request merged vs 200 OK
Yes, FS(13263) send out 482 request merged to my voip client. I guess, for some reason, FS doesn't respond to the REGISTER, and when the client start REGISTER again using another call-id, it merged the request to one. Anyone ever met this before? See the call-id and cseq below : recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:32.811280: REGISTER sip:voip.xxx.com SIP/2.0 CSeq: 208 REGISTER Content-Length: 0 recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:38.814237: REGISTER sip:voip.xxx.com SIP/2.0 CSeq: 210 REGISTER recv 631 bytes from udp/[69.131.94.250]:3270 at 12:40:48.821027: REGISTER sip:voip.xxx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:3270;rport;branch=z9hG4bKa9b7ba70783b617e9998dc4dd82eb3c5 From: sip:c...@voip.xxx.com:5090;tag=a9b7ba70783b617e9998dc4dd82eb3c5 To: cc sip:c...@voip.xxx.com:5090 Call-ID: a9b7ba70783b617e9998dc4dd82eb...@192.168.1.100 CSeq: 214 REGISTER Contact: sip:c...@192.168.1.100:3270;rinstance=1242647429 max-forwards: 70 expires: 300 Content-Length: 0 recv 629 bytes from udp/[69.131.94.250]:3270 at 12:40:52.841591: REGISTER sip:voip.xxx.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:3270;rport;branch=z9hG4bKb8c37e33defde51cf91e1e03e51657da From: cc sip:c...@voip.xxx.com: 5090;tag=b8c37e33defde51cf91e1e03e51657da To: cc sip:c...@voip.xxx.com:5090 Call-ID: b8c37e33defde51cf91e1e03e5165...@192.168.1.100 CSeq: 1 REGISTER Contact: sip:c...@192.168.1.100:3270;rinstance=1242650454 max-forwards: 70 expires: 300 sent 439 bytes to udp/[69.131.94.250]:3270 at 12:40:52.841767: SIP/2.0 482 Request merged Via: SIP/2.0/UDP 192.168.1.100 : 3270 ;rport = 3270 ;branch=z9hG4bKb8c37e33defde51cf91e1e03e51657da;received=69.131.94.250 From: cc sip:c...@voip.xxx.com: 5090;tag=b8c37e33defde51cf91e1e03e51657da To: cc sip:c...@voip.xxx.com:5090;tag=tjgccmtraDHFc Call-ID: b8c37e33defde51cf91e1e03e5165...@192.168.1.100 CSeq: 1 REGISTER Content-Length: 0 And I also noticed the CSeq if not continues, seems it lost some. but why the CSeq so big while the client directly logins to FS without any proxy and I don't think there is a loop? Anyway, don't know why FS does not respond to REGISTER sometimes. I updated FS to 13374, will see if it happen again. On May 19, 2009, at 3:37 AM, Brian West wrote: Is this in regards to FreeSWITCH or something else you're writing? /b On May 18, 2009, at 2:34 PM, dujinfang wrote: On register, sometimes my voip client got SIP/2.0 482 Request merged sometimes got 200 ok. 482 also means loop detected. my client only has one account logged in only one place, and no proxy, can I take 482 as 200 OK? Thanks. from RFC 3261: 8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] SIP/2.0 482 Request merged vs 200 OK
Please show me a pcap file to may email address because I can bet you FS/Sofia is doing it right 99% of the time. /b On May 18, 2009, at 6:55 PM, dujinfang wrote: es, FS(13263) send out 482 request merged to my voip client. I guess, for some reason, FS doesn't respond to the REGISTER, and when the client start REGISTER again using another call-id, it merged the request to one. Anyone ever met this before? See the call- id and cseq below : Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] SIP/2.0 482 Request merged vs 200 OK
I believe. capturing on tshark -i eth1 -w register.pcap udp port 5090 do you have further suggestions on the tshark filter? Thanks. On May 19, 2009, at 8:06 AM, Brian West wrote: Please show me a pcap file to may email address because I can bet you FS/Sofia is doing it right 99% of the time. /b On May 18, 2009, at 6:55 PM, dujinfang wrote: es, FS(13263) send out 482 request merged to my voip client. I guess, for some reason, FS doesn't respond to the REGISTER, and when the client start REGISTER again using another call-id, it merged the request to one. Anyone ever met this before? See the call-id and cseq below : Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] SIP/2.0 482 Request merged vs 200 OK
I never use tshark to capture... I use tcpdump -s0 -x port 5090 -w file.pcap /b On May 18, 2009, at 7:39 PM, dujinfang wrote: I believe. capturing on tshark -i eth1 -w register.pcap udp port 5090 do you have further suggestions on the tshark filter? Thanks. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Unable to successfully bridge calls to an external user
My suspicion is that the RTP traffic isn't traversing the NAT properly. You may have to configure the routers at both ends to forward the RTP packets to the correct destinations. There is a good discussion of NAT on the wiki. Situation: FS (10.0.0.12) - DMZ (124.254.81.250) - Internet - NAT (203.206.171.118) - Softphone (10.0.0.2) The problem is there's so much discussion of NAT that I'm not sure where to start. OK the problem is that I can't control the external user's router so I need a solution that works by only fixing the FS end. I've put my FS in the DMZ, but of course it's still got a local LAN IP address. Is there something I can configure to make FS realise that it _doesn't_ need to use NAT ? Whenever my softphones register to FS they register as UDP-NAT. Can I prevent that and make them register as regular UDP ? It would seem like they don't need to be in NAT mode since FS is in a DMZ, or do they ? I tried setting inbound-late-negotiation in my external (is this right ?) SIP profile and added proxy_media to my extension configurations in the dialplans, but this made no difference. It's possible that I haven't done this in the right spot or something. The other thing that looks promising is on http://wiki.freeswitch.org/wiki/External_profile which gives an example of a softphone registering to a NAT'd FS from outside on the internet (Switch with External Softphone example) which suggests I create a new external profile on a different port. I've done this and the user's softphone can register fine, but when he makes calls we still get no audio, presumably from lack of RTP data. I then tried adding in values for rtp-ip, sip-ip, ext-rtp-ip and ext-sip- ip on the new external profile to see if that made any difference but it didn't. Step 6 of the example says reference the caller from your FreeSWITCH system as: sofia/external5090/caller extension@x.x.x.x: 5090. I'm not sure what that means. Do I have to change something else to make it reference the caller by that external profile ? I figured it must be at least using that external profile because the phone is successfully registering on port 5090, but I'm not sure if I have to do something different to route incoming calls from the main external profile to the new 5090 one. I'm just not sure which NAT-related solution I'm supposed to be using. The External_profile wiki page example for the external softphone seems to fit my situation but didn't solve anything. The proxy_media solution seemed promising but had no real effect. It seems to me that the solution has something to do with having FS know that it's in a DMZ and that it doesn't need to do any NAT traversal, thereby making it think it's got a live internet IP and therefore only the external user would be using NAT traversal. I hope someone can give me some insight into which particular NAT- related solution I need because there seems to be dozens of ways to deal with this problem and I can't figure out which applies. ___ 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] No ringback for iPhone
I am not receiving any ringback when calling in from an iPhone. I receive
ringback when calling from other cell phones and land lines just not an
iPhone.
2009-05-18 20:28:04 [DEBUG] switch_core_state_machine.c:152
switch_core_standard_on_execute() sofia/external/3153836...@64.24.35.76
Execute set(ringback=${us-ring})
2009-05-18 20:28:04 [DEBUG] switch_core_session.c:1286
switch_core_session_exec() sofia/external/3153836...@64.24.35.76 Expanded
String set(ringback=%(2000, 4000, 440.0, 480.0))
2009-05-18 20:28:04 [DEBUG] mod_dptools.c:711 set_function()
sofia/external/3153836...@64.24.35.76 SET [ringback]=[%(2000, 4000, 440.0,
480.0)]
2009-05-18 20:28:04 [DEBUG] switch_core_state_machine.c:152
switch_core_standard_on_execute() sofia/external/3153836...@64.24.35.76
Execute set(transfer_ringback=local_stream://moh)
2009-05-18 20:28:04 [DEBUG] mod_dptools.c:711 set_function()
sofia/external/3153836...@64.24.35.76 SET
[transfer_ringback]=[local_stream://moh]
Any suggestions?
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