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Expert help needed debuging sip registration [logs included]
by Miguel Andrade 29 Oct '21

29 Oct '21

Some how I'm able to register sip phones and softphones but not as a freeswitch gateway. I've tried several combinations to no avail. My provider instructions are: · Endereço Registar (Host): sip.sapo.pt · Endereço Proxy (Outbound Proxy): proxy.sip.sapo.pt · Porta Proxy: 5070 · Username: +3513020xxxxx (*) · Password: [de 6 a 8 carateres] · Transporte Rede (Network Settings): UDP · SIP URL: +3513020xxxxx (*) I've entered this as username: +351xxx, password: xxx, proxy: proxy.sip.sapo.pt:5070 , from domain: sip.sapo.pt Doesn't work. I went to advanced and set transport to udp and outbound proxy as proxy.sip.sapo.pt:5070 didn't work wither. Tried all combinations I could but also no cigar. Can anyone help me debug this? Here are some fs_cli logs: 2021-10-15 15:41:46.615028 [NOTICE] sofia_reg.c:454 Registering ebf96e38-7592-43b7-82fa-4a9fd4c5facf nua_stack.c:569 nua_stack_signal() nua(0x7f6730005880): recv signal r_register nua_dialog.c:336 nua_dialog_usage_add() nua(0x7f6730005880): adding register usage nta.c:2670 nta_tpn_by_url() nta: selecting scheme sip nta.c:10831 outgoing_query_a() nta: for "proxy.sip.sapo.pt" query " proxy.sip.sapo.pt" A (cached) nta.c:10886 outgoing_answer_a() nta: proxy.sip.sapo.pt. IN A 194.65.2.241 tport.c:3274 tport_tsend() tport_tsend(0x7f6724195450) tpn = udp/ 194.65.2.241:5070 tport.c:4697 tport_by_addrinfo() tport_by_addrinfo(0x7f6724195450): not found by name udp/194.65.2.241:5070 tport.c:3609 tport_vsend() tport_vsend(0x7f6724195450): 642 bytes of 642 to udp/194.65.2.241:5070 nua_stack.c:528 nua_signal() nua(0x7f6730005880): sent signal r_register send 642 bytes to udp/[194.65.2.241]:5070 at 15:41:46.632220: ------------------------------------------------------------------------ REGISTER sip[image: image.gif]roxy.sip.sapo.pt:5070;transport=udp SIP/2.0 Via: SIP/2.0/UDP 92.222.246.205:5080;rport;branch=z9hG4bKc0QF2jmNeB8Qe Max-Forwards: 70 From: <sip:+351302020304@sip.sapo.pt>;tag=HjK0rD25pc27a To: <sip:+351302020304@sip.sapo.pt> Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1 CSeq: 42600661 REGISTER Contact: <sip:gw+ebf96e38-7592-43b7-82fa-4a9fd4c5facf@92.222.246.205:5080 ;transport=udp;gw=ebf96e38-7592-43b7-82fa-4a9fd4c5facf> Expires: 60 User-Agent: Grandstream HT802 1.0.3.2 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 nta.c:8312 outgoing_send() nta: sent REGISTER (42600661) to udp/ 194.65.2.241:5070 tport.c:4175 tport_pend() tport_pend(0x7f6724195450): pending 0x7f67241145e0 for udp/92.222.246.205:5080 (already 0) tport.c:2751 tport_wakeup_pri() tport_wakeup_pri(0x7f6724195450): events IN tport.c:2882 tport_recv_event() tport_recv_event(0x7f6724195450) tport.c:3219 tport_recv_iovec() tport_recv_iovec(0x7f6724195450) msg 0x7f672416c140 from (udp/92.222.246.205:5080) has 602 bytes, veclen = 1 recv 602 bytes from udp/[194.65.2.241]:5070 at 15:41:46.753181: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized 11030230330 Via: SIP/2.0/UDP 92.222.246.205:5080 ;received=92.222.246.205;branch=z9hG4bKc0QF2jmNeB8Qe;rport=5080 From: <sip:+351302020304@sip.sapo.pt>;tag=HjK0rD25pc27a To: <sip:+351302020304@sip.sapo.pt>;tag=9848f21803d1705ce6531d6ca2f Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1 CSeq: 42600661 REGISTER Content-Length: 0 WWW-Authenticate: Digest nonce="D734DDD02A93696100000000E09BC51B",realm=" ims.telecom.pt",algorithm=MD5,qop="auth" P-Charging-Vector: icid-value=3bnhe046p1ds2rk6ehcs2pg6hhoamfde9rpulo4623db2jmt;icid-generated-at=194.65.2.241;orig-ioi=internet tport.c:3037 tport_deliver() tport_deliver(0x7f6724195450): msg 0x7f672416c140 (602 bytes) from udp/194.65.2.241:5080/sip next=(nil) nta.c:3303 agent_recv_response() nta: received 401 Unauthorized 11030230330 for REGISTER (42600661) nta.c:3370 agent_recv_response() nta: 401 Unauthorized 11030230330 is going to a transaction nta.c:9584 outgoing_estimate_delay() nta_outgoing: RTT is 121.007 ms tport.c:4237 tport_release() tport_release(0x7f6724195450): 0x7f67241145e0 by 0x7f672454b980 with 0x7f672416c140 auth_digest.c:105 auth_digest_challenge_get() auth_digest_challenge_get(): got 6 nua_stack.c:271 nua_stack_event() nua(0x7f6730005880): event r_register 401 Unauthorized 11030230330 nua_stack.c:528 nua_signal() nua(0x7f6730005880): sent signal r_authenticate nua_stack.c:569 nua_stack_signal() nua(0x7f6730005880): recv signal r_authenticate auth_digest.c:225 auth_digest_a1() auth_digest_a1() has A1 = MD5(+351302020304:ims.telecom.pt:1q2w!Q) = 8038cbcec02479ac6c24e22c0d4d343a auth_digest.c:317 auth_digest_response() A2 = MD5(REGISTER:sip[image: image.gif]roxy.sip.sapo.pt:5070;transport=udp) auth_digest.c:339 auth_digest_response() auth_response: 72688ed841494a95193f7d115c409667 = MD5(8038cbcec02479ac6c24e22c0d4d343a[image: image.gif] 734DDD02A93696100000000E09BC51B:00000001:3UagCahoEjqJ2gIAACyybQ:auth:66f331 b6539f207dc0b673edce10bbbc) (qop=auth) nta.c:2670 nta_tpn_by_url() nta: selecting scheme sip nta.c:10831 outgoing_query_a() nta: for "proxy.sip.sapo.pt" query " proxy.sip.sapo.pt" A (cached) nta.c:10886 outgoing_answer_a() nta: proxy.sip.sapo.pt. IN A 194.65.2.241 tport.c:3274 tport_tsend() tport_tsend(0x7f6724195450) tpn = udp/ 194.65.2.241:5070 tport.c:4697 tport_by_addrinfo() tport_by_addrinfo(0x7f6724195450): not found by name udp/194.65.2.241:5070 tport.c:3609 tport_vsend() tport_vsend(0x7f6724195450): 920 bytes of 920 to udp/194.65.2.241:5070 send 920 bytes to udp/[194.65.2.241]:5070 at 15:41:46.753392: ------------------------------------------------------------------------ REGISTER sip[image: image.gif]roxy.sip.sapo.pt:5070;transport=udp SIP/2.0 Via: SIP/2.0/UDP 92.222.246.205:5080;rport;branch=z9hG4bKD9g83D5rBmyaa Max-Forwards: 70 From: <sip:+351302020304@sip.sapo.pt>;tag=HjK0rD25pc27a To: <sip:+351302020304@sip.sapo.pt> Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1 CSeq: 42600662 REGISTER Contact: <sip:gw+ebf96e38-7592-43b7-82fa-4a9fd4c5facf@92.222.246.205:5080 ;transport=udp;gw=ebf96e38-7592-43b7-82fa-4a9fd4c5facf> Expires: 60 User-Agent: Grandstream HT802 1.0.3.2 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Authorization: Digest username="+351302020304", realm="ims.telecom.pt", nonce="D734DDD02A93696100000000E09BC51B", cnonce="3UagCahoEjqJ2gIAACyybQ", algorithm=MD5, uri="sip[image: image.gif]roxy.sip.sapo.pt:5070;transport= udp", response="72688ed841494a95193f7d115c409667", qop=auth, nc=00000001 Content-Length: 0 nta.c:8312 outgoing_send() nta: sent REGISTER (42600662) to udp/ 194.65.2.241:5070 tport.c:4175 tport_pend() tport_pend(0x7f6724195450): pending 0x7f67242e5a90 for udp/92.222.246.205:5080 (already 0) tport.c:2751 tport_wakeup_pri() tport_wakeup_pri(0x7f6724195450): events IN tport.c:2882 tport_recv_event() tport_recv_event(0x7f6724195450) tport.c:3219 tport_recv_iovec() tport_recv_iovec(0x7f6724195450) msg 0x7f67242d2000 from (udp/92.222.246.205:5080) has 602 bytes, veclen = 1 recv 602 bytes from udp/[194.65.2.241]:5070 at 15:41:46.871252: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized 1103023032F Via: SIP/2.0/UDP 92.222.246.205:5080 ;received=92.222.246.205;branch=z9hG4bKD9g83D5rBmyaa;rport=5080 From: <sip:+351302020304@sip.sapo.pt>;tag=HjK0rD25pc27a To: <sip:+351302020304@sip.sapo.pt>;tag=9848f2180cea805ce6531d8948d Call-ID: 8ab00b1b-ec4d-4564-b148-26fcb82622e1 CSeq: 42600662 REGISTER Content-Length: 0 WWW-Authenticate: Digest nonce="47DFEAEB2A936961000000000782EF64",realm=" ims.telecom.pt",algorithm=MD5,qop="auth" P-Charging-Vector: icid-value=3bnhe046p1ds2rk6ehcs2pg6hhoamfde9rpulo462blb2jmt;icid-generated-at=194.65.2.241;orig-ioi=internet tport.c:3037 tport_deliver() tport_deliver(0x7f6724195450): msg 0x7f67242d2000 (602 bytes) from udp/194.65.2.241:5080/sip next=(nil) nta.c:3303 agent_recv_response() nta: received 401 Unauthorized 1103023032F for REGISTER (42600662) nta.c:3370 agent_recv_response() nta: 401 Unauthorized 1103023032F is going to a transaction nta.c:9584 outgoing_estimate_delay() nta_outgoing: RTT is 117.905 ms tport.c:4237 tport_release() tport_release(0x7f6724195450): 0x7f67242e5a90 by 0x7f672410d3b0 with 0x7f67242d2000 auth_digest.c:105 auth_digest_challenge_get() auth_digest_challenge_get(): got 6 nua_client.c:1223 nua_base_client_check_restart() nua(0x7f6730005880): bad credentials, clearing them nua_stack.c:271 nua_stack_event() nua(0x7f6730005880): event r_register 401 Unauthorized 1103023032F nua_stack.c:528 nua_signal() nua(0x7f6730005880): sent signal r_authenticate nua_stack.c:569 nua_stack_signal() nua(0x7f6730005880): recv signal r_authenticate nua_stack.c:271 nua_stack_event() nua(0x7f6730005880): event r_register 904 Operation has no matching challenge nua_dialog.c:395 nua_dialog_usage_remove_at() nua(0x7f6730005880): removing register usage 2021-10-15 15:41:46.854983 [ERR] sofia_reg.c:2470 ebf96e38-7592-43b7-82fa-4a9fd4c5facf Failed Registration with status Operation has no matching challenge [904]. failure #1

2 1

Changing device on one leg improves the audio recorded from another leg?
by David P 29 Oct '21

29 Oct '21

We had this bizarre experience today: A user on this device... Dell Latitude 5580 i7-7600U CPU Windows 10 Education OS 19042.685 ...made verto calls in which he heard clicks and pops every few 100ms from our virtual agent on another leg, and those audio defects also appear in the mp4 recording. When he then switched to this device *on the same desk* using the same network... Dell Latitude 5590 i5-8350U CPU Windows 10 Education OS 17763.1821 ...there were no audio defects. Playing audio from other sources on the problematic device revealed no problems with its speaker. At least one of the failed calls used srflx candidates, not any relaying. How could changing the device on leg A affect the audio recording that apparently comes from leg B? If it's relevant: - We use "jitterbuffer_msec=5p:100p" - xml_cdr logs for both legs show seemingly high values for some aspects like these... - <skip_packet_count>187</skip_packet_count> <jitter_packet_count>1620</jitter_packet_count>

2 1

Trying to get short hostname
by Antony Stone 28 Oct '21

28 Oct '21

Hi. I'm using FreeSwitch 1.10.7 as a client registering to other PBXs as well as a SIP server itself. I am setting the user-agent string in the SIP profile, such as: <param name="user-agent-string" value="FS $${CFGver} $${hostname}"/> The variable $${CFGver}} is a version string I'm definiing in vars.xml This is working, but it gives me the FQDN of the machine as $${hostname}} and I really only want the machine name without the domain. I've found https://freeswitch.org/confluence/display/FREESWITCH/Regular+Expression and https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+regex and I've tried to use them to cut the domain off the value of $${hostname} but with no success: In vars.xml: <X-PRE-PROCESS cmd="set" data="shorthostname=${regex($${hostname}|([^\.]*).*| %1)}"/> This results in the variable shorthostname being assigned the value: %{regex(abcd.efgh.tld|([^\.]*).*|%1)} where abcd.efgh.tld is the server's FQDN. Instead of doing that in vars.xml and creating the variable shorthostname, I tried putting the regex into the SIP profile: <param name="user-agent-string" value="FS $${CFGver} ${regex($${hostname}| ([^\.]*).*|%1)}"/> This ends up with precisely the same result. The ${regex(....)} function simply isn't being executed. Given that the machine's hostname is also available from the system command "hostname -s" I decided to try that instead: In vars.xml: <X-PRE-PROCESS cmd="set" data="shorthostname=${system(hostname -s)}"/> Or in the SIP profile definition: <param name="user-agent-string" value="FS $${CFGver} ${system(hostname -s)}"/> These both produce the output: FS 0.65 ${system(hostname -s)} In other words, the ${system(....)} command isn't working any more than ${regex(....)} did. What am I doing wrong please? How can I get the machine's short hostname into a user-agent string for registration to a server? Thanks, Antony. -- Perfection in design is achieved not when there is nothing left to add, but rather when there is nothing left to take away. - Antoine de Saint-Exupery Please reply to the list; please *don't* CC me.

2 2

Postgress BDR insert failed for "complete" table
by Abdirahman Osman 28 Oct '21

28 Oct '21

Hello, I am using postgres with BDR bidirectional replication which requires primary keys for the freeswitch tables. So far so good , except freeswitch is complaining about one table here is the error 2021-10-26 23:40:47.772796 95.47% [DEBUG] mod_pgsql.c:784 Query (insert into complete values (0,'hash_remote', 'list', '', '', '', '', '', '', '', '', 'dagobah-api-gw-1-2')) returned PGRES_FATAL_ERROR 2021-10-26 23:40:47.772796 95.47% [ERR] mod_pgsql.c:551 Error executing query: ERROR: column "complete_uuid" is of type uuid but expression is of type integer LINE 1: insert into complete values (0,'hash_remote', 'list', '', ''... What I noticed is that freeswitch is doing an INSERT INTO command with no column names specified. I checked the source code and found this to be the only case for the "complete" table whereas the other tables when INSERT INTO is used column names are specified. Here is the structure of my complete table with the new column complete_uuid as primary key. CREATE TABLE :myschema.complete ( complete_uuid uuid DEFAULT public.gen_random_uuid() PRIMARY KEY, sticky INTEGER, a1 VARCHAR(128), a2 VARCHAR(128), a3 VARCHAR(128), a4 VARCHAR(128), a5 VARCHAR(128), a6 VARCHAR(128), a7 VARCHAR(128), a8 VARCHAR(128), a9 VARCHAR(128), a10 VARCHAR(128), hostname VARCHAR(256) ); Any suggestions? Thanks Abdirahman A. Osman

2 3

Issue with mod_event_multicast in 1.10.7
by Koldo Aingeru 28 Oct '21

28 Oct '21

Hi, I just updated a test server from FS 1.10.6 to 1.10.7 on Debian 10 (apt update/upgrade from the official repo) and without changing anything on my config I got a buffer overflow when running "/usr/bin/freeswitch -u freeswitch -g freeswitch -c -nonat ": *** buffer overflow detected ***: /usr/bin/freeswitch terminated Aborted I tried the original modules.xml that comes with the package and it starts up without problems. Doing a bit of trial & error with my modules.conf.xml I noticed that the problema is with the mod_event_multicast module. Loading it from modules.conf.xml or by doing “load mod_event_multicast” result in a crash. This is the current module: # sha1sum /usr/lib/freeswitch/mod/mod_event_multicast.so 4daee6fa8ae8ed59c5ebc32393cb7357fead8198 /usr/lib/freeswitch/mod/mod_event_multicast.so Overwriting the module with the 1.10.6 one seems to work: freeswitch(a)pbx.mydomain <mailto:freeswitch@pbx.mydomain>.com> load mod_event_multicast +OK Reloading XML +OK 2021-10-28 14:21:23.038592 96.17% [CONSOLE] switch_loadable_module.c:1768 Successfully Loaded [mod_event_multicast] 2021-10-28 14:21:23.038592 96.17% [NOTICE] switch_loadable_module.c:389 Adding API Function 'multicast_peers' but even with the module loaded the events aren't being sent to my other server, my event_multicast_conf.xml is this: <configuration name="event_multicast.conf" description="Multicast Event"> <settings> <param name="address" value="225.1.1.1"/> <param name="port" value="4242"/> <param name="bindings" value="CUSTOM sofia::register"/> <param name="ttl" value="1"/>     <param name="psk" value=“pbx.mydomain.com"/> </settings> </configuration> I don’t see any multicast traffic being sent from this server nor the " Propagating registration for….” message. Any idea what can be happening? Thanks! ======================================== Koldo Aingeru Marcos Fdez. Ingeniería de Sistemas Sarenet S.A.U. 944209470 Parque Tecnológico, Edificio 103 48170 Zamudio, Bizkaia www.sarenet.es ========================================

1 0

Updating caller id with attended transfer while phone still ringing
by Rahman Duran 28 Oct '21

28 Oct '21

Hi, When doing attended transfer (A -> B -> attended transfer -> C) C only updates its caller id display when it answers the call. Is there anyway to update caller id on C, when B transfer the call while its still ringing? The problem is we have some Cisco and Samsung SIP phones that does not have a real "Blind Transfer" support. If you want to transfer without talking to "C", you just press the transfer button quickly, while C starts to ring. So I want to send the updated caller id to "C" after "B" transferred but before "C" answers. I have Yeastar PBX (asterisk based) works just like that but I can't figure out how to do it with freeswitch. Regards, Rahman Duran

1 0

problems compiling 1.10.7
by John Covici 27 Oct '21

27 Oct '21

Hi. I am compiling freeswitch under gentoo and getting the following error: /bin/sh /usr/src/freeswitch/libs/apr/libtool --silent --mode=compile gcc -g -O2 -pthread -g -O2 -DHAVE_CONFIG_H -D_REENTRANT -D_GNU_SOURCE -I./include -I/usr/src/freeswitch/libs/apr/include/arch/unix -I./include/arch/unix -I/usr/src/fr\eeswitch/libs/apr/include -o threadproc/unix/signals.lo -c threadproc/unix/signals.c && touch threadproc/unix/signals.lo In file included from ./include/apr_pools.h:39, from /usr/src/freeswitch/libs/apr/include/arch/unix/../apr_private_common.h:24, from /usr/src/freeswitch/libs/apr/include/arch/unix/apr_private.h:881, from /usr/src/freeswitch/libs/apr/include/arch/unix/apr_arch_threadproc.h:18, from threadproc/unix/signals.c:18: ./include/apr_general.h:33: warning: "NSIG" redefined 33 | #define NSIG __DARWIN_NSIG | In file included from /usr/include/sys/wait.h:36, from ./include/apr.h:131, from /usr/src/freeswitch/libs/apr/include/arch/unix/apr_arch_threadproc.h:17, from threadproc/unix/signals.c:18: /usr/include/signal.h:184: note: this is the location of the previous definition 184 | # define NSIG _NSIG | In file included from ./include/apr_pools.h:39, from /usr/src/freeswitch/libs/apr/include/arch/unix/../apr_private_common.h:24, from /usr/src/freeswitch/libs/apr/include/arch/unix/apr_private.h:881, from /usr/src/freeswitch/libs/apr/include/arch/unix/apr_arch_threadproc.h:18, from threadproc/unix/signals.c:18: ./include/apr_general.h:33:14: error: `__DARWIN_NSIG' undeclared here (not in a function) 33 | #define NSIG __DARWIN_NSIG | ^~~~~~~~~~~~~ threadproc/unix/signals.c:127:20: note: in expansion of macro `NSIG' 127 | #define APR_NUMSIG NSIG | ^~~~ threadproc/unix/signals.c:136:39: note: in expansion of macro `APR_NUMSIG' 136 | static const char *signal_description[APR_NUMSIG]; | ^~~~~~~~~~ make[2]: *** [/usr/src/freeswitch/libs/apr/build/apr_rules.mk:201: threadproc/unix/signals.lo] Error 1 Anyone have a clue? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici(a)ccs.covici.com

1 0

AUDIO RTP REPORTS ERROR: [Remote Address Error!] -- [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
by Vishal Pai 27 Oct '21

27 Oct '21

Hello Everyone I am using webrtc client sip.js + freeswitch 1.10.2 installed from src. Whenever the first call comes to sip.js it gets disconnected with error AUDIO RTP REPORTS ERROR: [Remote Address Error!] -- [INCOMPATIBLE_DESTINATION] Here is the log file attached. Is there any bug in the freeswitch version ? Or am I using the wrong configuration? Note there is no NAT on the server, it DO droplet which is directly connected to public IP. Any help will be highly appreciated. Thanks.

3 2

Buld freeswitch - Visual Studio
by Gregor Nanger 26 Oct '21

26 Oct '21

Hi! I wanted to build the latest version of FS in Visual Studio 2017. I get following errors: Error MSB4044 The "SwitchVersionTask" task was not given a value for the required parameter "revision". FreeSwitchCoreLib C:\freeswitch-1.10.7\w32\switch_version.props 128 Error The "SignFile" task was not given a value for the required parameter "SigningTarget". Setup.2017 C:\freeswitch-1.10.7\w32\Setup\Setup.2017.wixproj 1025 Since FreeSwitchCoreLib is not built, I got a lot of other errors. Can someone help me with this? BR, Gregor

1 0

Re: [Freeswitch-users] Internal extension and external CELL phone - revisited
by Brian West 25 Oct '21

25 Oct '21

ignore_early_media=true will be required here. On Mon, Oct 25, 2021 at 11:49 AM Sean Devoy via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Sean Devoy <sdevoy(a)bizfocused.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 25 Oct 2021 16:13:56 +0000 > Subject: Internal extension and external CELL phone - revisited > > Hi, > > > > With Avi’s help I have the dial string corrected to: > > Dial an internal extension. Delay 10 seconds, then dial external number > (which happens to be a cell phone) > > The goal is to set the call timeout to keep the voicemail in FreeSwitch. > (For this discussion let’s ignore the variability of time to make a > connection to a cell phone.) > > > > After 10 second delay, FreeSwitch dials the external number just fine. > Then the external leg gets a > > switch_core_media.c:8980 sofia/external/14108089999 Set 2833 dtmf receive > payload to 101 > > It immediately hangs up the internal extension with a > > switch_ivr_originate.c:3716 Hangup sofia/internal/220@68.143.96.162:1458 > [CS_CONSUME_MEDIA] [LOSE_RACE] > > > > Note, the cell phone never gets answered by the recipient. Sadly, the call > does go to the cell voicemail. > > > > Here are my questions: > > 1 – Is that because the Cell Carrier has actually “answered” the call and > is now routing it to the phone? > > 2 – Can I even go to voicemail on FS after that event or is the call > “answered”? > > > > Regards, > > Sean > > > > ---------- Forwarded message ---------- > From: Sean Devoy via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 25 Oct 2021 09:49:37 -0700 (PDT) > Subject: [Freeswitch-users] Internal extension and external CELL phone - > revisited > _________________________________________________________________________ > > 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 -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>

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