November 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

S3 Upload Issue
by Danish Daud 22 Nov '21

22 Nov '21

Hi Users, We have a scenario, in which we are uploading recording file on "S3" (AWS) after completion of a call through python script. We encountered error in uploading recording file on S3 for some calls and for some it worked fine. After debugging the issue we have this scenario: Currently when the calls hangup after being answered (before we get the below mentioned logs in fs_cli), the script returns us an error (logs attached). But if we get the below mentioned [Debug] logs and call hangup after these, no error occurs and script successfully upload the file to s3. We have not encountered that issue before. [Debug Logs] 2021-11-19 02:58:08.748629 [DEBUG] switch_ivr_async.c:1427 Record session sample rate: 8000 -> 8000 2021-11-19 02:58:08.748629 [DEBUG] switch_core_media_bug.c:970 Attaching BUG to sofia/external/+18162394364 2021-11-19 02:58:08.768629 [DEBUG] switch_ivr_async.c:1717 No silence detection configured; assuming start of speech 2021-11-19 02:58:09.588628 [DEBUG] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-11-19 02:58:09.588628 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2021-11-19 02:59:02.468628 [DEBUG] sofia.c:7406 Channel sofia/internal/3175520010@3.210.46.94:12000 entering state [calling][0] 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7406 Channel sofia/internal/3175520010@3.210.46.94:12000 entering state [ready][200] 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:7416 Remote SDP: v=0 o=Z 0 2 IN IP4 44.227.145.86 s=Z c=IN IP4 44.227.145.86 t=0 0 m=audio 57578 RTP/AVP 0 106 9 8 3 101 98 a=rtpmap:106 opus/48000/2 a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:98 telephone-event/48000 a=fmtp:98 0-16 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5656 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [opus:106:48000:20:0:2]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5601 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5517 Set telephone-event payload to 101@8000 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5860 Set telephone-event payload to 101@8000 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:5918 sofia/internal/3175520010@3.210.46.94:12000 Set 2833 dtmf send payload to 101 recv payload to 101 2021-11-19 02:59:02.928627 [DEBUG] sofia.c:8427 Processing updated SDP 2021-11-19 02:59:02.928627 [DEBUG] switch_core_media.c:8657 Audio params are unchanged for sofia/internal/3175520010@3.210.46.94:12000. ------------------------------------------------------------------------------------------------------------------------------------------ [Upload Error logs] 2021-11-19 02:47:03.348630 [ERR] mod_python.c:317 Error calling python script 2021-11-19 02:47:03.348630 [ERR] mod_python.c:165 Python Error by calling script "s3_upload": <class 'boto3.exceptions.S3UploadFailedError'> Message: Failed to upload /tmp/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav to storage.rvm.sas2.co/3175520010_12028997788_fa1b6644-9779-42bb-a830-6bf51381484f.wav: An error occurred (BadDigest) when calling the PutObject operation (reached max retries: 4): The Content-MD5 you specified did not match what we received. Exception: None Traceback (most recent call last) File: "/usr/local/freeswitch/scripts/s3_upload.py", line 22, in fsapi File: "/usr/local/lib/python2.7/dist-packages/boto3/s3/inject.py", line 131, in upload_file File: "/usr/local/lib/python2.7/dist-packages/boto3/s3/transfer.py", line 287, in upload_file ------------------------------------------------------------------------------------------------------------------------------------ Any help in this regard is highly appreciated. Regards, Danish Daud

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mod_xml_cdr & centos 7 via yum install
by Glen Christie 21 Nov '21

21 Nov '21

So I installed freeswitch via yum on an update to date centos 7 system using the following instructions from https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 yum install -y https://files.freeswitch.org/repo/yum/centos-release/freeswitch-release-rep… epel-release yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-* mod_xml_cdr doesn't seem to be installed by default, it's not in /usr/lib64/freeswitch/mod autoload_configs has xml_cdr.conf.xml and I uncommented <load module="mod_xml_cdr"/> in modules.conf.xml but doing a load mod_xml_cdr in the cli just produces the below 2021-11-17 14:43:30.825959 98.40% [CRIT] switch_loadable_module.c:1750 Error Loading module /usr/lib64/freeswitch/mod/mod_xml_cdr.so **/usr/lib64/freeswitch/mod/mod_xml_cdr.so: cannot open shared object file: No such file or directory** Is mod_xml_cdr depreciated now? I cant seem to find any info on whether it is or not.

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Preserving recovery information during restart
by TTI 20 Nov '21

20 Nov '21

I have a FS running and due to a hardware problem need to reboot the machine. I would like to preserve recovery information (so calls can resume). Is it possible to use the shutdown command to gracefully shutdown FS without clearing the recovery data? The manual shows shutdown options as: shutdown [asap|asap restart|cancel|elegant|now|restart|restart asap|restart elegant] and I'm wondering if the 'now' parameters does this? If not I assume I have to kill -SIGKILL the process, to ensure that recovery data is not cleared out?

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[Not sending re-invite on 5XX message]
by Bilal Abbasi 17 Nov '21

17 Nov '21

Hi Users, We are using freeswitch to send calls to carrier through SRV records, what happens is if carrier sends 503 response FreeSWITCH thinks of that call as rejection and sends to other available IPs under SRV record. That's not leading us to non 5xx error, as the carrier told us that if you get 503 don't expect any other response from other available IPs. So is there anyway i can restrict FS to not send re-invites in case of a 503 response? Regards Bilal Abbasi

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ESL asynchronous use
by Telium Technical Support 17 Nov '21

17 Nov '21

I'm trying to figure out asynchronous use of the ESL (from C). If I call esl_connect_timeout with a timeout of 0, what happens if the connection is not made immediately? Is there some callback function indicating success/failure? If the esl connection is lost (FreeSwitch crashes), is there a callback function indicating the lost connection? Or do I need to issue a status/keepalive to ensure function succeeds? Looking a bit at the esl.c code I'm wondering if the above are synchronous only.(polling for status)

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How to retrieve all waiting events via ESL in C/C++
by Telium Technical Support 16 Nov '21

16 Nov '21

I am trying to retrieve all waiting events via ESL, using the esl_recv_event_timed function. The documentation doesn't offer much help. My code below is intended to retrieve all events waiting. But it doesn't work...I just gather an increasing number of headers (always appending onto the last), and the body is the same, for each event. As if the events are not dequeuing. As well, events I expect (eg HEARTBEAT) are not appearing. So it seems I'm not really iterating through the list of waiting events: QStringList body; esl_event_t *save_event = NULL; while (true) { esl_status_t connectionStatus = esl_recv_event_timed(&m_EslHandle, 0, 0, &save_event); if (connectionStatus == ESL_SUCCESS) { if (save_event) { esl_event_header_t* nextHeader = save_event->headers; while ((nextHeader != NULL) && (nextHeader != save_event->last_header)) { headerList << TL_KeyValuePair(nextHeader->name,nextHeader->value); nextHeader = nextHeader->next; } body = QString(m_EslHandle.last_sr_event->body).split("\n",Qt::SkipEmptyParts); } else break; } else { return false; } } Can someone offer a simple code (using C ESL) on how to retrieve all waiting events?

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fs_cli command syntax
by Telium Technical Support 15 Nov '21

15 Nov '21

I was using the cli trying to enable events. After much playing and searching I discovered that I must prefix my command with slash; eg: /event plain all Why is this? I can't find any reference to why this command needs a slash prefix?

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Bridge to other FS server has no audio until DTMF
by Avi Marcus 14 Nov '21

14 Nov '21

I started a new thread in case anyone muted it... it wasn't simply a network issue. It seems the bridging occurs and dialplan processes, but no media flows - until DTMF from the A-leg. Call flow: PSTN (via carrier) to freeswitch A -> media and IVR -> freeswitch B. Calls directly from carrier to Freeswitch B are fine. Calls from a different carrier to Freeswitch A -> media and IVR -> Freeswitch B are also fine. So it sounds like a carrier/unique SIP/RTP issue, but since FS is in the media path, it's an FS issue... I actually mcguyvered this right now with a queue_dtmf before the bridge, to force the audio stream to update. Here's the log on freeswitch B: EXECUTE [depth=0] sofia/external/972581234567(a)172.123.123.123 log(DEBUG class chosen: 1234567) 2021-10-07 09:16:24.343175 [DEBUG] mod_dptools.c:1879 class chosen: 1234567 EXECUTE [depth=0] sofia/external/972581234567(a)172.123.123.123 javascript(conference/lookupAndJoinConference.js 1234567) EXECUTE [depth=0] sofia/external/972581234567(a)172.123.123.123 playback(class/hold-wait-teacher.wav) 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567(a)172.123.123.123 entering state [completed][200] 2021-10-07 09:16:24.363379 [DEBUG] sofia.c:7406 Channel sofia/external/ 972581234567(a)172.123.123.123 entering state [ready][200] 2021-10-07 09:16:24.363379 [DEBUG ] switch_ivr_play_say.c:1486 Codec Activated L16@8000hz 1 channels 20ms 2021-10-07 09:16:34.903283 [DEBUG ] switch_rtp.c:7793 Correct audio ip/port confirmed. 2021-10-07 09:16:34.923190 [DEBUG] switch_rtp.c:8038 RTP RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [INFO] switch_channel.c:522 RECV DTMF 3:2080 2021-10-07 09:16:34.923190 [DEBUG] mod_dptools.c:2389 Digit 3 2021-10-07 09:16:37.143169 [DEBUG ] switch_ivr_play_say.c:1931 done playing file /usr/share/freeswitch/sounds/en/us/matthew/class/hold-wait-teacher.wav You can see a 10 second gap between call ready 200 and correct audio/ip and file done playing (it's a 2 second file), and this doesn't happen automatically, only when I choose to press something. Any ideas as to the root cause of this? -Avi Marcus ---------- Forwarded message --------- From: Avi Marcus <avi(a)avimarcus.net> Date: Wed, Oct 6, 2021 at 3:32 PM Subject: Bridge to other FS server has no audio ??? To: FreeSWITCH Users Help <FreeSWITCH-users(a)lists.freeswitch.org> Any ideas on why a call doesn't have media? It used to work, but I think my upstream changed his SDP again. - FreeSWITCH Server A - call comes in and bypass_media bridges to FS server B. Media works. - FreeSWITCH Server A - call comes in and bridges to FS server B (not on bypass). Media works. - FreeSWITCH Server A - call comes in, gets answered, then bridges to FS server B. Call looks OK, but no media is flowing (I don't hear anything, PCAPs just have SIP, and there isn't 80kbps network traffic). All the same codecs are set in the json cdrs (PCMU). FS server B is to join a conference if that matters. I was assuming it had to do with codecs, but setting absolute_codec_string to PCMU doesn't make any difference in the logs - it's already always PCMU. I have NO clue what further could cause this other than codecs, which seem to be fine. Any ideas please? -Avi Marcus

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Use of esl_recv_event
by Telium Technical Support 13 Nov '21

13 Nov '21

I'm looking at how to use the esl_recv_event function in esl.c and the docs are missing some details. Could someone tell me: 1. Do I have to lock a mutex before accessing this function (since it dequeues an event) 2. The third parameter is a ptr to a ptr to a struct. So, do I have to allocate a struct before calling (to get the data)? Or will the lib allocate the struct and I have to free it? Comments from esl.c: /*! \brief Poll the handle's socket until an event is received or a connection error occurs \param handle Handle to poll \param check_q If set to 1, will check the handle queue (handle->race_event) and return the last event from it \param[out] save_event If this is not NULL, will return the event received */ ESL_DECLARE(esl_status_t) esl_recv_event(esl_handle_t *handle, int check_q, esl_event_t **save_event); /*!

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Looking for libesl, headers, etc.
by Telium Technical Support 12 Nov '21

12 Nov '21

I built a Debian 9 based FreeSwitch+FusionPBX from packages. I now want to create a simple C++ program talking to FS using the ESL. I can't find libesl on the system, nor header files. Is this available as a separate download package? Do I have to DL the whole FS source, compile, and pull that out?

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