- FreeSWITCH-users - FreeSWITCH Mailing Lists

Infolink, ServerPronto, ColoPronto, Etc <YES ITS IS OFF TOPIC>
by Ken Rice 02 Aug '11

02 Aug '11

Just a word of warning... There is a company in Florida that operates under many company names including but not limited to InfoLink, ServerPronto, ColoPronto, and others... This Company makes it IMPOSSIBLE to close an account... I had colocation (rented a dedicated server) from them for several months, during the entire time, the bandwidth was never right, altho they promised several TB of transfer, bandwidth was such that promised transfer amounts were never achievable... I secured separate Colo, issued an order to cancel the account by opening a ticket through their secured web portal... These idiots continue to attempt to charge my credit card to the point where I had to cancel my card and order a new one... And til this day they continue to try and collect and refuse to refund any money they collected after the cancel order... Avoid these guys like the plague unless you want to jump through hoops K

1 0

mod_rtmp question
by Madovsky 02 Aug '11

02 Aug '11

I use freeswitch in ODBC mode. in sofia endpoint if I type "sofia status profile internal" I get all registrations from all nodes but in rtmp endpoint "rtmp status profile internal reg" I get registrations of the local node. is it a normal behaviour ? if yes which rtmp command can give the same result of sofia with ODBC ? Thanks Franck

2 2

How to use ESL event
by David Ma 02 Aug '11

02 Aug '11

Hello All, I have some problem with receiving and handling ESL events. My code is like: void handle_eslswitch_event_plain(esl_handle_t *handle, esl_event_t *event) { ... ... TRACE_DEBUG("ESL event [%d] %s: UUID: %s", event->event_id, fs_get_event_header(event, "Event-Name"), uniqueid); print_event(event); ... ... } void handle_event(esl_handle_t *handle, esl_event_t *last_event) { ... ... if (!strcasecmp(type, "text/event-plain")) { handle_eslswitch_event_plain(handle, handle->last_ievent); } ... ... } int main() { ... ... //receive from event socket if (handle.last_event) handle_event(&handle, handle.last_event); ... ... Usually I receive correct events. But I occasionally receive incorrect event. See below the 1st is the normal event and the 2nd is the wrong event: <2011-07-29 14:25:29> [DEBUG] ESL event [8] CHANNEL_HANGUP_COMPLETE: UUID: f320e4a6-db23-46d0-8d89-9957eccbd4c9 <2011-07-29 14:25:29> [DEBUG] RECV EVENT Event-Name: CHANNEL_HANGUP_COMPLETE Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72 FreeSWITCH-Hostname: fs01 FreeSWITCH-IPv4: 10.1.1.46 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2011-07-29 14:25:29 However, usually after failure in executing "hangup" and getting "-ERR ..." in the event->last_sr_reply, I have wrong event, but not always (an -ERR returned from executing "hangup" doesn't always result in a wrong event). Here it is: <2011-07-29 14:25:29> [NOTICE] -ERR invalid session id [f320e4a6-db23-46d0-8d89-9957eccbd4c9] <2011-07-28 15:55:23> [DEBUG] ESL event [0] : UUID: <2011-07-28 15:55:23> [DEBUG] RECV EVENT Content-Length: 6485 Content-Type: text/event-plain Event-Name: CHANNEL_HANGUP Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72 FreeSWITCH-Hostname: fs01 FreeSWITCH-IPv4: 10.1.1.46 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2011-07-28%2015%3A55%3A23 In the above case, both Event ID and UUID are invalid. I think i need more information regarding how to use the data elements in ESL event, especially how to use the following: char last_reply[1024]; /*! Las command reply when called with esl_send_recv */ char last_sr_reply[1024]; /*! Last event received. Only populated when **save_event is NULL */ esl_event_t *last_event; /*! Last event received when called by esl_send_recv */ esl_event_t *last_sr_event; /*! This will hold already processed events queued by esl_recv_event */ esl_event_t *race_event; /*! Events that have content-type == text/plain and a body */ esl_event_t *last_ievent; My questions, what event should I check if I am interested in only channel/call related events, and how? Do I check handle.last_event, or handle.last_ievent, or what? Please kindly advice. Thanks, D.Ma

1 0

ESL: not bein able to determine when SIP gateway is down when originating a call
by Anton VG 02 Aug '11

02 Aug '11

I'm trying to catch an error, in case I would dial wrong (non existent) gateway (intentionally!) I'm running ESL outbound listener, subscribing to all events, if I do bgapi 'originate' to a live gateway - there are normal events flow, and I can track what is happening. But if I issue originate to a gateway, which is not configured or simply down - there are no any events fired. I only have an error on FS console 2011-05-14 20:58:13.072927 [ERR] mod_sofia.c:4044 Invalid Gateway following by 2011-05-14 20:58:13.072927 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] But HOW to catch the given in ESL? sofia.c seems just does not have event code for that cases if (profile_name && !profile_found) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "No Such Profile '%s'\n", profile_name); >status = SWITCH_STATUS_FALSE; } logically there should be a proper way to determine that gateway is bad in my ESL dialplan, by catching the proper event/reply/whatever, For the moment i did trick: esl.api('sofia status gateway GatewayWhichIsDown') When in production, and there is more than a single route, there will be plenty of cases, when you dial a bad gateway, so there should be a way for ESL dialplan to determine that a gateway is not callable for a moment, the reason WHY and to retry with another one. The trick above is bad, since: 1. blocking api query, before evey single gateway call attempt. 2. Gateway maybe known in UP state, but the state is stale, in dial in fact will go to DOWN gateway. So, ESL dialplan will screw in that case Any clue?

4 4

Various question - CallerID / Call transfer via REFER and CDR
by Weigel, Stefan 02 Aug '11

02 Aug '11

Hi all, I'm currently stuck with some issues I can't figure out. I'm using FreeSWITCH Version 1.0.head (git-db5f504 2011-07-17 17-00-38 -0400), phone devices are Polycom Soundpoint IP 560. 1. When receiving a call I change some of the incoming Caller-ID names. It's working like a charm. But if I do a call forward (it's a 'Consultative transfer'-Party A dials party C's number and talks privately with party C after the call is answered, and then completes the transfer or hangs up) I only see the party A's number/name on the display C. Is there a possibility to get the original caller-ID/-name displayed on the phone of Party C ? 2. When doing a 'Consultative transfer', I only get one CDR entry using mod_cdr_csv. For example calling from my cellphone 0179 12345678 to extension 615, then doing a call transfer from 615 to 803 I get an entry in CDR with calling_id = 615, destination_number = 803. Is it possible to trigger the creation of a CDR entry ? I tried using process_cdr but only one entry get's logged. In cdr_csv.conf I'm using logging of a-leg, for billing / call statistic. Thanks in advance and best regards Stefan

1 0

Re: [Freeswitch-users] Sip Headers advice (Not parsing properly)
by Dissident 02 Aug '11

02 Aug '11

Problem solved!!! It seems that this doc is misleading -> http://wiki.freeswitch.org/wiki/Channel_Variables#Channel_Variable_Manipula… not all sip headers are gonna be parsed only the ones which start with X- or P- the rest are not parsed because they are not supposed to exist, however in my case they do and the only solution I've found to parse my CISCO headers is modifying the mod_sofia, file sofia.c line 6515, I added the last OR condition. } else if (!strncasecmp(un->un_name, "X-", 2) || !strncasecmp(un->un_name, "P-", 2) || !strncasecmp(un->un_name, "Cisco", 5)) { and now I can retrieve this header from the dialplan just like this: ${sip_h_Cisco-Guid}"/> Thanks to Swk. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sip-Headers-advice-Not-parsin… Sent from the freeswitch-users mailing list archive at Nabble.com.

5 5

ptime
by michael knop 02 Aug '11

02 Aug '11

Hi all, I’m using FreeSWITCH Version 1.0.head (git-0fc8050 2011-07-31 22-14-06 -0500) and I’m dealing with a problem <http://pastebin.freeswitch.org/16913> that sounds like the one that is described in the Jira bug report <http://jira.freeswitch.org/browse/FS-2003>. Is the patch to reset the timestamp detector every time FreeSWITCH gets a new sdp or reply to invite still active? Thanks, micha

1 1

Various question - CallerID / Call transfer via REFER and CDR
by Weigel, Stefan 02 Aug '11

02 Aug '11

Hi all, I'm currently stuck with some issues I can't figure out. I'm using FreeSWITCH Version 1.0.head (git-db5f504 2011-07-17 17-00-38 -0400), phone devices are Polycom Soundpoint IP 560. 1. When receiving a call I change some of the incoming Caller-ID names. It's working like a charm. But if I do a call forward (it's a 'Consultative transfer'-Party A dials party C's number and talks privately with party C after the call is answered, and then completes the transfer or hangs up) I only see the party A's number/name on the display C. Is there a possibility to get the original caller-ID/-name displayed on the phone of Party C ? 2. When doing a 'Consultative transfer', I only get one CDR entry using mod_cdr_csv. For example calling from my cellphone 0179 12345678 to extension 615, then doing a call transfer from 615 to 803 I get an entry in CDR with calling_id = 615, destination_number = 803. Is it possible to trigger the creation of a CDR entry ? I tried using process_cdr but only one entry get's logged. In cdr_csv.conf I'm using logging of a-leg, for billing / call statistic. Thanks in advance and best regards Stefan

1 0

Audio issues when FS as PSTN/SBC Gateway for OpenSIPS
by Deon Vermeulen 02 Aug '11

02 Aug '11

Hi List My Current setup is as follows. PSTN -- Sangoma A500 bri -- FS -- OpenSIPS DID: bridges called number to ext@opensips My Phones are located on our office private LAN, while FS and OpenSIPS are both on Public Network with Public IPs. There is NAT between the IP Phones and OpenSIPS/FS. I am using non standard SIP ports as well as not defined any rules for RTP on the firewall. I have a many to one NAT scenario on the Firewall. OpenSIPS is setup to not Proxy any media between the phones and FS. Media flows directly between FS and phones. When I make calls to PSTN, audio is Crystal clear on both sides of the call. When I receive calls from the PSTN I sometimes have audio, sometimes none and most of the time one way (to PSTN). I have done quite a lot of sip trace debugging in fs_cli and I really can't see any problems in the traces. I have also enabled inbound-proxy-media in all sip profiles, just to make sure I don't miss it somewhere in the call flow. I have been digging around in the wiki, forums as well as some archived mails wrt FS and NAT, but I don't 100% understand how FS handles NAT by default in the External.xml profile. I'm also not 100% sure how FS will handle this type of scenario as no Phones are registered to it and it also has to pass SIP and SDP info with OpenSIPS and not directly with the Phones. Any advise and or guidance will really be appreciated. Thank you very much. Kind Regards Deon Vermeulen

1 0

mod_radius_cdr does not work
by fieldpeak 02 Aug '11

02 Aug '11

Dear All, i'm trying to use mod_radius_cdr, and FS successfully loaded it, but when i make a call, it show error below on console, could anyone can help? Thanks in advance. 2011-08-02 14:45:32.786095 [ERR] mod_radius_cdr.c:242 failed adding Freeswitch-Src: 1002 *debug level log on console:* freeswitch@freeswitch> 2011-08-02 14:45:32.546104 [DEBUG] sofia.c:7030 IP 222.128.70.10 Rejected by acl "domains". Falling back to Digest auth. 2011-08-02 14:45:32.546104 [WARNING] sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for [1001(a)124.193.106.104] from ip 222.128.70.10 2011-08-02 14:45:32.786095 [DEBUG] sofia.c:7030 IP 222.128.70.10 Rejected by acl "domains". Falling back to Digest auth. 2011-08-02 14:45:32.786095 [NOTICE] switch_channel.c:897 New Channel sofia/internal/1002(a)124.193.106.104 [d8030a1e-552b-4a32-b7f0-175653b65eb7] 2011-08-02 14:45:32.786095 [DEBUG] sofia.c:5084 Channel sofia/internal/ 1002(a)124.193.106.104 entering state [received][100] 2011-08-02 14:45:32.786095 [DEBUG] sofia.c:5095 Remote SDP: v=0 o=- 186895328 186895344 IN IP4 192.168.200.100 s=eyeBeam c=IN IP4 192.168.200.100 t=0 0 m=audio 6200 RTP/AVP 18 100 6 0 8 3 98 97 5 102 101 a=rtpmap:100 speex/16000 a=rtpmap:98 ilbc/8000 a=rtpmap:97 speex/8000 a=rtpmap:102 l16/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : 21373DA4 8CC12811 192.168.200.100 6200 2011-08-02 14:45:32.786095 [DEBUG] sofia.c:5258 (sofia/internal/ 1002(a)124.193.106.104) State Change CS_NEW -> CS_INIT 2011-08-02 14:45:32.786095 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_INIT 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1002(a)124.193.106.104) State INIT 2011-08-02 14:45:32.786095 [DEBUG] mod_sofia.c:85 sofia/internal/ 1002(a)124.193.106.104 SOFIA INIT 2011-08-02 14:45:32.786095 [DEBUG] mod_sofia.c:125 (sofia/internal/ 1002(a)124.193.106.104) State Change CS_INIT -> CS_ROUTING 2011-08-02 14:45:32.786095 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1002(a)124.193.106.104) State INIT going to sleep 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_ROUTING 2011-08-02 14:45:32.786095 [DEBUG] switch_channel.c:1821 (sofia/internal/ 1002(a)124.193.106.104) Callstate Change DOWN -> RINGING 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:373 (sofia/internal/1002(a)124.193.106.104) State ROUTING 2011-08-02 14:45:32.786095 [DEBUG] mod_sofia.c:148 sofia/internal/ 1002(a)124.193.106.104 SOFIA ROUTING 2011-08-02 14:45:32.786095 [DEBUG] mod_radius_cdr.c:156 [mod_radius_cdr] Entering my_on_routing 2011-08-02 14:45:32.786095 [ERR] mod_radius_cdr.c:242 failed adding Freeswitch-Src: 1002 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:373 (sofia/internal/1002(a)124.193.106.104) State ROUTING going to sleep 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] sofia.c:5084 Channel sofia/internal/ 1002(a)124.193.106.104 entering state [terminated][487] 2011-08-02 14:45:36.185722 [DEBUG] switch_channel.c:2739 (sofia/internal/ 1002(a)124.193.106.104) Callstate Change RINGING -> HANGUP 2011-08-02 14:45:36.185722 [NOTICE] sofia.c:5806 Hangup sofia/internal/ 1002(a)124.193.106.104 [CS_ROUTING] [ORIGINATOR_CANCEL] 2011-08-02 14:45:36.185722 [DEBUG] switch_channel.c:2755 Send signal sofia/internal/1002(a)124.193.106.104 [KILL] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_HANGUP 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:575 (sofia/internal/1002(a)124.193.106.104) State HANGUP 2011-08-02 14:45:36.185722 [DEBUG] mod_sofia.c:452 sofia/internal/ 1002(a)124.193.106.104 Overriding SIP cause 487 with 487 from the other leg 2011-08-02 14:45:36.185722 [DEBUG] mod_sofia.c:458 Channel sofia/internal/ 1002(a)124.193.106.104 hanging up, cause: ORIGINATOR_CANCEL 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1002(a)124.193.106.104 Standard HANGUP, cause: ORIGINATOR_CANCEL 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:575 (sofia/internal/1002(a)124.193.106.104) State HANGUP going to sleep 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1002(a)124.193.106.104) State Change CS_HANGUP -> CS_REPORTING 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_REPORTING 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:635 (sofia/internal/1002(a)124.193.106.104) State REPORTING 2011-08-02 14:45:36.185722 [DEBUG] mod_radius_cdr.c:424 [mod_radius_cdr] Entering my_on_reporting 2011-08-02 14:45:36.185722 [ERR] mod_radius_cdr.c:463 failed adding Freeswitch-Hangupcause: 487 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:635 (sofia/internal/1002(a)124.193.106.104) State REPORTING going to sleep 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1002(a)124.193.106.104) State Change CS_REPORTING -> CS_DESTROY 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1326 Session 2 (sofia/internal/1002(a)124.193.106.104) Locked, Waiting on external entities 2011-08-02 14:45:36.185722 [NOTICE] switch_core_session.c:1344 Session 2 (sofia/internal/1002(a)124.193.106.104) Ended 2011-08-02 14:45:36.185722 [NOTICE] switch_core_session.c:1346 Close Channel sofia/internal/1002(a)124.193.106.104 [CS_DESTROY] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/1002(a)124.193.106.104) Callstate Change HANGUP -> DOWN 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_DESTROY 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/1002(a)124.193.106.104) State DESTROY 2011-08-02 14:45:36.185722 [DEBUG] mod_sofia.c:363 sofia/internal/ 1002(a)124.193.106.104 SOFIA DESTROY 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1002(a)124.193.106.104 Standard DESTROY 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/1002(a)124.193.106.104) State DESTROY going to sleep Regards, Charles

1 0