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Rum python script from dialplan
by Ricardo Martinez 28 Jun '21

28 Jun '21

Hi I’m trying to run a script in python from the freeswitch dialplan. This scripts has calls to a local REDIS DB which I need to query every time a call arrives. So far I was able to compile freeswitch with mod_python, I even can run a test.py script from the fs_cli like this: freeswitch@reverse> python test 2021-06-25 16:36:36.007177 [NOTICE] mod_python.c:213 Invoking py module: test 2021-06-25 16:36:36.027120 [INFO] switch_cpp.cpp:1465 test Hello But when try to add in the test.py the “import redis” line in order to work with my redis db I’m getting : freeswitch@reverse> python test 2021-06-25 16:37:17.847161 [NOTICE] mod_python.c:213 Invoking py module: test 2021-06-25 16:37:17.847161 [ERR] mod_python.c:261 Error reloading module 2021-06-25 16:37:17.847161 [ERR] mod_python.c:165 Python Error by calling script "test": <type 'exceptions.ImportError'> Message: No module named redis Exception: None Traceback (most recent call last) File: "/usr/share/freeswitch/scripts/test.py", line 1, in <module> Maybe I’m not getting correctly the use of pythons scripts?.. Can I use a script that calls another module?... Is possible? Or maybe there is a better way to do this?. Thanks! Ricardo

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Run a python script from dialplan.
by Ricardo Martinez 25 Jun '21

25 Jun '21

Hi I’m trying to run a script in python from the freeswitch dialplan. This scripts has calls to a local REDIS DB which I need to query every time a call arrives. So far I was able to compile freeswitch with mod_python, I even can run a test.py script from the fs_cli like this: freeswitch@reverse> python test 2021-06-25 16:36:36.007177 [NOTICE] mod_python.c:213 Invoking py module: test 2021-06-25 16:36:36.027120 [INFO] switch_cpp.cpp:1465 test Hello But when try to add in the test.py the “import redis” line in order to work with my redis db I’m getting : freeswitch@reverse> python test 2021-06-25 16:37:17.847161 [NOTICE] mod_python.c:213 Invoking py module: test 2021-06-25 16:37:17.847161 [ERR] mod_python.c:261 Error reloading module 2021-06-25 16:37:17.847161 [ERR] mod_python.c:165 Python Error by calling script "test": <type 'exceptions.ImportError'> Message: No module named redis Exception: None Traceback (most recent call last) File: "/usr/share/freeswitch/scripts/test.py", line 1, in <module> Maybe I’m not getting correctly the use of pythons scripts?.. Can I use a script that calls another module?... Is possible? Or maybe there is a better way to do this?. Thanks! *Ricardo*

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Re: [Freeswitch-users] rtp-timeout-sec VS media_timeout
by David P 25 Jun '21

25 Jun '21

Hi Allan, I don't know if the media_timeout=300 behavior you saw is a bug or not, but I wanted to add my own observation of weirdness about hangup cause MEDIA_TIMEOUT... I just noticed a conference end abruptly after one leg spoke for 5 minutes. The logs aren't clear why this happened, but it seems that <param name="rtp-timeout-sec" value="300"/> in our sip_profiles/internal.xml is the reason -- it seems that because the *other* leg of the conference remained silent, the RTP timeout was reached. I couldn't find any confluence pages about MEDIA_TIMEOUT by googling. On Fri, Jan 8, 2021 at 1:08 AM < freeswitch-users-request(a)lists.freeswitch.org> wrote: > > ---------- Forwarded message ---------- > From: Allan Kristensen <ak(a)hejdu.dk> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 6 Jan 2021 19:37:33 +0100 > Subject: [Freeswitch-users] rtp-timeout-sec VS media_timeout > We had some problems with "hanging channels" for our webrtc clients (via > kamailio). > To solve the problem I tried to use "media_timeout" setting but it didn't > really work. So I tried the deprecated "rtp-timeout-sec" and this actually > works fine? > > Not working: > <param name="media_timeout" value="300"/> > > Working: > <param name="rtp-timeout-sec" value="300"/> > > How to reproduce: Make a call using webrtc and just close browser window, > after some time freeswitch should close the channel because of missing rtp. > Instead it just keeps writing: "switch_rtp.c:853 No audio stun for a long > time!" forever... > > Anyway, it works now....just curious why...a typo or bug? > > /Allan > > Using: > FreeSWITCH Version 1.10.5-release+git~20200818T185121Z~25569c1631~64bit >

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Copy or Extract Contact Parameters from A-leg to B-leg
by Chris Wallace 23 Jun '21

23 Jun '21

We are working on implementing STIR/SHAKEN and have run into a snag when passing calls through our FreeSwitch LCR which sits between our call feature server and our SBC's. Our call feature server is passing parameters in the Contact for attestation-info and origination-id. It is intended that our SBC's will extract this data from the Contact and use it to generate an identity header before leaving our network. Ideally this information would be passed via X-Header(s) but unfortunately this is what we have to work with at the moment. We have tried to use both the "sip_invite_contact_params" and "sip_contact_params" channel variables but neither of them seem to get any data from the Contact within the inbound INVITE. We had success testing with "sip_contact_user" and dumping that data to an X-Header but we can't seem to get FreeSwitch to recognize or dump the other parameters. When debugging an active call and dumping the channel variable data for each leg of the call we don't see any of the additional parameters that are in the Contact listed in the channel variables that are dumped for either call leg's UUID. *Contact in INVITE into FreeSwitch* Contact: <sip:15558675309@x.x.x.x:5060>;isup-oli=00;attestation-info=1;origination-id=1000;verstat=0 *sip_contact_user Dump to X-Header:* X-ContactUser: 15558675309 *FreeSwitch Logging:* EXECUTE [depth=0] sofia/core/15558675309@x.x.x.x:5060 export(sip_h_X-ContactUser=15558675309) 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactUser]=[15558675309] EXECUTE [depth=0] sofia/core/15558675309@x.x.x.x:5060 export(sip_h_X-ContactParam=) 2021-06-18 15:24:10.079576 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [sip_h_X-ContactParam]=[UNDEF]

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Outgoing calls with a AVM FritzBox 7490
by Janne Heß 23 Jun '21

23 Jun '21

Hello everyone, I'm kind of lost with setting up FS to connect to my FritzBox 7490. The goal is to use FS with spandsp to send and receive Faxes. Now, most things work in my setup. FS can successfully register as a SIP client and I can call the configured number from my mobile phoneand the call is routed to the FS0 virtual modem. The problem is outgoing calls. Using the ATD command on the virtual modem to call any external phone does not work. Internal FritzBox-configured numbers don't work either. Using Wireshark I found that the FritzBox replies with 488 Not Acceptable Here. Looking around the internet, this seems to be related to the codec configuration. I doubt this is the problem in my case since enabling more codecs doesn't help and incoming calls with the same codec restrictions work. So I'm assuming FS sends an INVITE package that the FritzBox does not like for some reason. I played around with the From field but that just results in the FritzBox not replying at all. Does anyone know what might be going on here or is there someone with a working example? I have attached the INVITE package and the response package. 192.168.0.130 is the FritzBox, 192.168.0.133 is the FS host (with firewall disabled). Thank you in advance and best regards Janne Frame 183405: 1278 bytes on wire (10224 bits), 1278 bytes captured (10224 bits) on interface -, id 0 Ethernet II, Src: RealtekU_ff:ff:ff (52:54:00:ff:ff:ff), Dst: AVMAudio_ff:ff:ff (7c:ff:4d:ff:ff:ff) Internet Protocol Version 4, Src: 192.168.0.133, Dst: 192.128.0.130 User Datagram Protocol, Src Port: 5060, Dst Port: 5060 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:0123456789@192.168.0.130:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.0.133;rport;branch=z9hG4bKcU2Qc6yc0cZ3p Max-Forwards: 70 From: "FSModem" <sip:hylafaxtel@192.168.0.130>;tag=j2K0XQKgXBFmN To: <sip:0123456789@192.168.0.130:5060> Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216 [Generated Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216] CSeq: 37632110 INVITE Contact: <sip:gw+fritzbox@192.168.0.133:5060;transport=udp;gw=fritzbox> User-Agent: FreeSWITCH-mod_sofia/1.10.6-release.12~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Authorization: Digest username="hylafaxtel", realm="fritz.box", nonce="XXXXX", algorithm=MD5, uri="sip:0123456789@192.168.0.130:5060", response="XXXXX" Content-Type: application/sdp Content-Disposition: session Content-Length: 225 X-FS-Support: update_display,send_info Remote-Party-ID: "FSModem" <sip:FS0@129.143.6.130>;party=calling;screen=yes;privacy=off Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): FreeSWITCH 1624344691 1624344692 IN IP4 192.168.0.133 Session Name (s): FreeSWITCH Connection Information (c): IN IP4 192.168.0.133 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 27112 RTP/AVP 102 101 Media Attribute (a): rtpmap:102 L16/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ptime:20 [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216] [Generated Call-ID: 751E5AC1C59CE6B0(a)192.168.0.130] [I removed most Call-IDs for brevity] Frame 183406: 837 bytes on wire (6696 bits), 837 bytes captured (6696 bits) on interface -, id 0 Ethernet II, Src: AVMAudio_ff:ff:ff (7c:ff:4d:ff:ff:ff), Dst: RealtekU_ff:ff:ff (52:54:00:ff:ff:ff) Internet Protocol Version 4, Src: 192.168.0.130, Dst: 192.168.0.133 User Datagram Protocol, Src Port: 5060, Dst Port: 5060 Session Initiation Protocol (488) Status-Line: SIP/2.0 488 Not Acceptable Here Message Header Via: SIP/2.0/UDP 192.168.0.133;rport=5060;branch=z9hG4bKcU2Qc6yc0cZ3p From: "FSModem" <sip:hylafaxtel@192.168.0.130>;tag=j2K0XQKgXBFmN To: <sip:0123456789@192.168.0.130:5060>;tag=CDE9429F6B0A3BC3 Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216 [Generated Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216] CSeq: 37632110 INVITE Warning: 399 0.0.0.0 "successful but result empty" User-Agent: FRITZ!OS Content-Type: application/sdp Content-Length: 361 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): user 12041099 12041099 IN IP4 129.143.6.130 Session Name (s): call Connection Information (c): IN IP4 192.168.0.130 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 7080 RTP/AVP 8 0 2 102 100 99 97 101 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:102 G726-32/8000 Media Attribute (a): rtpmap:100 G726-40/8000 Media Attribute (a): rtpmap:99 G726-24/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): fmtp:97 mode=30 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): rtcp:7081 [Generated Call-ID: 596BDCD5EDA427D7(a)192.168.0.130] [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216] [I removed most Call-IDs for brevity]

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Re: [Freeswitch-users] UDP port Operation not permitted
by David Villasmil 17 Jun '21

17 Jun '21

This usually happens because either the IP address in the profile is wrong, or maybe fs is running with the wrong user. Verify the IP address both in the profile and on vars, and try running fs from the cli as root and see if anything changes. On Thu, 17 Jun 2021 at 08:42, Wayne Chang via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Wayne Chang <wayne_w_chang(a)yahoo.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>, Mike > Jerris <mike(a)freeswitch.org> > Cc: > Bcc: > Date: Thu, 17 Jun 2021 07:42:11 +0000 (UTC) > Subject: Re: [Freeswitch-users] UDP port Operation not permitted > Hi Mike, > > what could be the issue? I am sure 10.201.21.140 has port 5060 listening. > I tested with SIPp and the target 10.201.21.140 could receive INVITE > properly. > > The issue only happened when running following command from fs_cli. > > freeswitch@freeswitch> originate sofia/external/ > PaymentIVR2-34117(a)10.201.21.140 1002 > Thanks > > Wayne > > > On Monday, June 14, 2021, 10:45:50 AM PDT, Mike Jerris < > mike(a)freeswitch.org> wrote: > > > tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not > permitted with (s=35 UDP/10.201.21.140:5060) > > write to network failing. > > On Jun 14, 2021, at 11:42 AM, Wayne Chang via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> wrote: > > > *From: *Wayne Chang <wayne_w_chang(a)yahoo.com> > *Subject: **UDP port Operation not permitted* > *Date: *June 13, 2021 at 11:41:45 PM MDT > *To: *"freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > > > Hi, > > I am using the fs_cli command to originate a call to a media server. > However I am receiving the 503 Service Unavailable. I have checked the > media server listening on port 5060. I am able to use sip client to send > the invite. Can you help me understand why when using the originate command > it failed? > > Thanks. Here is the log. > > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:585 > (sofia/external/PaymentIVR2-34117) Running State Change CS_CONSUME_MEDIA > (Cur 1 Tot 29) > soa_static.c:1028 soa_sdp_mode_set() soa_sdp_mode_set(0x7f1347ffc5c0, > (nil), ""): called > soa_static.c:1445 offer_answer_step() soa_static(0x7f133c0412b0, > soa_generate_offer): storing local description > soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, > [(nil)], [0x7f1347ffe720], [0x7f1347ffe718]) called > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 > (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA > nta.c:2694 nta_tpn_by_url() nta: selecting scheme sip > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 > (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA going to sleep > tport.c:3285 tport_tsend() tport_tsend(0x7f133c004e80) tpn = UDP/ > 10.201.21.140:5060 > tport.c:4071 tport_resolve() tport_resolve addrinfo = 10.201.21.140:5060 > tport.c:4708 tport_by_addrinfo() tport_by_addrinfo(0x7f133c004e80): not > found by name UDP/10.201.21.140:5060 > tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not > permitted with (s=35 UDP/10.201.21.140:5060) > tport.c:3521 tport_send_msg() tport_vsend returned -1 > nta.c:8542 outgoing_print_tport_error() nta: INVITE (37269853): Operation > not permitted (1) with UDP/[10.201.21.140]:5060 > nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call > state changed: init -> calling, sent offer > soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, > [0x7f1347ffe730], [0x7f1347ffe738], [(nil)]) called > nua_stack.c:269 nua_stack_event() nua(0x7f136c038fd0): event i_state > INVITE sent > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event r_invite 503 > Service Unavailable > nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call > state changed: calling -> init > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_state 503 > Service Unavailable > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_terminated > 503 Service Unavailable > > > > > > ---------- Forwarded message ---------- > From: Wayne Chang via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>, Mike > Jerris <mike(a)freeswitch.org> > Cc: > Bcc: > Date: Thu, 17 Jun 2021 00:42:49 -0700 (PDT) > Subject: Re: [Freeswitch-users] UDP port Operation not permitted > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [Freeswitch-users] UDP port Operation not permitted
by Mike Jerris 17 Jun '21

17 Jun '21

tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not permitted with (s=35 UDP/10.201.21.140:5060) write to network failing. > On Jun 14, 2021, at 11:42 AM, Wayne Chang via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: Wayne Chang <wayne_w_chang(a)yahoo.com> > Subject: UDP port Operation not permitted > Date: June 13, 2021 at 11:41:45 PM MDT > To: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> > > > Hi, > > I am using the fs_cli command to originate a call to a media server. However I am receiving the 503 Service Unavailable. I have checked the media server listening on port 5060. I am able to use sip client to send the invite. Can you help me understand why when using the originate command it failed? > > Thanks. Here is the log. > > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:585 (sofia/external/PaymentIVR2-34117) Running State Change CS_CONSUME_MEDIA (Cur 1 Tot 29) > soa_static.c:1028 soa_sdp_mode_set() soa_sdp_mode_set(0x7f1347ffc5c0, (nil), ""): called > soa_static.c:1445 offer_answer_step() soa_static(0x7f133c0412b0, soa_generate_offer): storing local description > soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [(nil)], [0x7f1347ffe720], [0x7f1347ffe718]) called > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA > nta.c:2694 nta_tpn_by_url() nta: selecting scheme sip > 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA going to sleep > tport.c:3285 tport_tsend() tport_tsend(0x7f133c004e80) tpn = UDP/10.201.21.140:5060 > tport.c:4071 tport_resolve() tport_resolve addrinfo = 10.201.21.140:5060 > tport.c:4708 tport_by_addrinfo() tport_by_addrinfo(0x7f133c004e80): not found by name UDP/10.201.21.140:5060 > tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not permitted with (s=35 UDP/10.201.21.140:5060) > tport.c:3521 tport_send_msg() tport_vsend returned -1 > nta.c:8542 outgoing_print_tport_error() nta: INVITE (37269853): Operation not permitted (1) with UDP/[10.201.21.140]:5060 > nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: init -> calling, sent offer > soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [0x7f1347ffe730], [0x7f1347ffe738], [(nil)]) called > nua_stack.c:269 nua_stack_event() nua(0x7f136c038fd0): event i_state INVITE sent > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event r_invite 503 Service Unavailable > nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: calling -> init > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_state 503 Service Unavailable > nua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_terminated 503 Service Unavailable >

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channel variable for time left when using sched_hangup ?
by Julien Terrasson 14 Jun '21

14 Jun '21

Hello, Does anybody know if there is a way to capture the time remaining before a channel is hanged_up with sched_hangup ? Is there somewhere a channel variable that can reflect this information ? Regards, Julien

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UDP port Operation not permitted
by Wayne Chang 14 Jun '21

14 Jun '21

Hi, I am using the fs_cli command to originate a call to a media server. However I am receiving the 503 Service Unavailable. I have checked the media server listening on port 5060. I am able to use sip client to send the invite. Can you help me understand why when using the originate command it failed? Thanks. Here is the log. 2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:585 (sofia/external/PaymentIVR2-34117) Running State Change CS_CONSUME_MEDIA (Cur 1 Tot 29)soa_static.c:1028 soa_sdp_mode_set() soa_sdp_mode_set(0x7f1347ffc5c0, (nil), ""): calledsoa_static.c:1445 offer_answer_step() soa_static(0x7f133c0412b0, soa_generate_offer): storing local descriptionsoa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [(nil)], [0x7f1347ffe720], [0x7f1347ffe718]) called2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIAnta.c:2694 nta_tpn_by_url() nta: selecting scheme sip2021-06-13 22:08:10.682603 [DEBUG] switch_core_state_machine.c:663 (sofia/external/PaymentIVR2-34117) State CONSUME_MEDIA going to sleeptport.c:3285 tport_tsend() tport_tsend(0x7f133c004e80) tpn = UDP/10.201.21.140:5060tport.c:4071 tport_resolve() tport_resolve addrinfo = 10.201.21.140:5060tport.c:4708 tport_by_addrinfo() tport_by_addrinfo(0x7f133c004e80): not found by name UDP/10.201.21.140:5060tport.c:3663 tport_send_fatal() tport_vsend(0x7f133c004e80): Operation not permitted with (s=35 UDP/10.201.21.140:5060)tport.c:3521 tport_send_msg() tport_vsend returned -1nta.c:8542 outgoing_print_tport_error() nta: INVITE (37269853): Operation not permitted (1) with UDP/[10.201.21.140]:5060nua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: init -> calling, sent offersoa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x7f133c0412b0, [0x7f1347ffe730], [0x7f1347ffe738], [(nil)]) callednua_stack.c:269 nua_stack_event() nua(0x7f136c038fd0): event i_state INVITE sentnua_stack.c:359 nua_application_event() nua: nua_application_event: enteringnua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event r_invite 503 Service Unavailablenua_session.c:4135 signal_call_state_change() nua(0x7f136c038fd0): call state changed: calling -> initnua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_state 503 Service Unavailablenua_stack.c:271 nua_stack_event() nua(0x7f136c038fd0): event i_terminated 503 Service Unavailable Wayne

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How to build Sofia-sip stack 32bit version for android
by Raveendra Battala 09 Jun '21

09 Jun '21

Hi, I am trying to build sofia-sip-1.12.11 or sofia-sip-1.13.3 32bit version for android but build failed always with the following error. Could you share information me whether Sofia-sip stack supports arm 32bit version build for android? export AR=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-ar export AS=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-as export CC=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-gcc export CXX=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-g++ export LD=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-ld export RANLIB=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-ranlib export STRIP=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-strip export NM=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/arm-linux-androideabi-nm export PATH=$NDK/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin:$PATH ~/WORK/SofiaSIP/sofia-sip-1.12.11$ ./configure --host=arm-linux-androideabi --build=arm-linux-androideabi checking build system type... Invalid configuration `arm-linux-androideabi': system `androideabi' not recognized configure: error: /bin/bash ./config.sub arm-linux-androideabi failed Thanks & Regards, Raveendra NOTICE AND DISCLAIMER This email contains Vectone information, which may be privileged or confidential. It's meant only for the individual(s) or entity named above. If you're not the intended recipient, note that disclosing, copying, distributing or using this information is prohibited. If you've received this email in error, please let me know immediately on the email address above. We monitor our email system, and may record your emails. Thank you.

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