July 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Presence issue. How packet should look?
by Igor Olhovskiy 26 Jul '16

26 Jul '16

Hi! I’m trying to subscribe to Freeswitch with this packet SUBSCRIBE sip:*55@master.rufan.at SIP/2.0 Record-Route: <sip:10.0.20.70:5060;ftag=jz8i26dy0f;lr=on> Via: SIP/2.0/UDP 212.232.26.232:5060 ;branch=z9hG4bKbb4c.eafe0e8f9ec4b28fb6e4056a64c702ed.0 Via: SIP/2.0/UDP 192.168.88.59:37600 ;received=176.37.91.15;branch=z9hG4bK-rzfuku5tc99o;rport=37600 Route: <sip:10.0.20.70:5060;ftag=jz8i26dy0f;lr=on> From: <sip:607@master.rufan.at>;tag=jz8i26dy0f To: <sip:*55@master.rufan.at;user=phone>;tag=qBuxIeW1MNWF Call-ID: 313435373534383931393431313632-94bfcwdyrzz2 CSeq: 610 SUBSCRIBE Max-Forwards: 69 User-Agent: snom760/8.7.5.35 Contact: <sip:607@176.37.91.15:37600;line=ukipitb6>;reg-id=1 Event: dialog Accept: application/dialog-info+xml Expires: 80 Content-Length: 0 Path: <sip:212.232.26.232:5060;lr;received=sip:176.37.91.15:37600> X-AUTH-IP: 176.37.91.15 Actually I get Accepted (202) But with NOTIFY from freeswitch I got this NOTIFY sip:607@176.37.91.15:37600;line=ukipitb6 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.71;rport;branch=z9hG4bKS5ZKag4N6XXFK Route: <sip:10.0.20.70:5060>;ftag=jz8i26dy0f;lr=on Record-Route: <sip:10.0.20.70:5060>;ftag=jz8i26dy0f;lr=on Max-Forwards: 70 From: <sip:*55@master.rufan.at;user=phone>;tag=qBuxIeW1MNWF To: <sip:607@master.rufan.at>;tag=jz8i26dy0f Call-ID: 313435373534383931393431313632-94bfcwdyrzz2 CSeq: 297327351 NOTIFY Contact: <sip:*55@10.0.20.71:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Event: dialog Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=80 Content-Type: application/dialog-info+xml Content-Length: 152 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="8" state="full" entity="sip:*55@master.rufan.at"> </dialog-info> Problem is, XML data is empty, cause actually there is no registrations for *55 extension. Why it’s not sending «Unavail» or smth. like this? What is the correct packet for SUBSCRIBE? Or internal profile? Internal profile is nonce-ttl [60] auth-calls [true] inbound-reg-force-matching-username [true] auth-all-packets [false] ext-rtp-ip [10.0.20.71] ext-sip-ip [10.0.20.71] rtp-timeout-sec [300] rtp-hold-timeout-sec [1800] tls-verify-policy [all] multiple-registrations [contact] enable-timer [false] dbname [share_presence] send-presence-on-register [true] inbound-codec-negotiation [greedy] NDLB-force-rport [safe] challenge-realm [auto_to] outbound-proxy [10.0.20.70] track-calls [true] nat-options-ping [true] liberal-dtmf [true] all-reg-options-ping [true] force-publish-expires [true] unregister-on-options-fail [true] user-agent-string [FreeSWITCH] log-auth-failures [true] forward-unsolicited-mwi-notify [false] context [public] rfc2833-pt [101] sip-port [5060] dialplan [XML] dtmf-duration [2000] inbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] outbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] rtp-timer-name [soft] rtp-ip [10.0.20.71] sip-ip [10.0.20.71] hold-music [local_stream://default] apply-nat-acl [nat.auto] aggressive-nat-detection [true] apply-inbound-acl [domains] local-network-acl [localnet.auto] record-path [/usr/local/freeswitch/recordings] record-template [${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}] manage-presence [true] presence-probe-on-register [true] manage-shared-appearance [true] tls [false] -- Best regards, Igor

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Skill based routing in mod_callcenter
by Ashwin Rath 26 Jul '16

26 Jul '16

Hi All Can i implement skill based routing in mod_callcenter ? can it be done by modifying the "tier level" of an agent in a queue ? -- Ashwin Kumar Rath

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Hangup Cause for a ringing destination leg
by Daniel Schäfer 26 Jul '16

26 Jul '16

Dear Freeswitch Users and Developers, I'm a bit desperate. I googled and tried multiple days last weeks and I can't find a solution for my (hopefully simple) problem: In short: how do I set the hangup cause for a ringing destination leg? Long story: I'm struggling to get a certain call-group behaviour: Incoming call --> freeswitch --> User A & [delayed] Group A Group A is meant to be something like a Hunt Group, so that the incoming call can be answered by someone else if User A is not available. It is not for sure that someone on Group A will answer the incoming call. I want to record the missed call only to User A. My main struggling point: If the Incoming call is aborted, the call log of User A and of all members of Group A is set to a missed call. And depending on my differently attempted implementations I can also have two missed calls for User A (one from initial ringing, one from group ringing). So I thought that I need to modify the hangup cause, but I can't get it working at all (I can modify the hangup cause for the Incoming call only). Could you point me to the right wording and directions? I thought I'll need two bridges: 1. bridge for User A, onetimeout set hangup cause for the destination leg 2. bridge for User A and Group A, onetimeout set hangup causes for destination legs But how do I do the hangup cause for ringing destination leg? Best Regards Daniel Schäfer

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How to Call Forwarding to new extension if called extension is Busy?
by sachindeshpande 26 Jul '16

26 Jul '16

Hello All, I am trying to implement the following scenario on Freeswitch under Windows. I am freeswitch version is 1.5. I have 3 clients registered to freeswitch : 1000, 1001, 1002 *Issue 1* If my first registered client say 1000 is busy, I want new calls to be directed to other registered user say 1001. If 1000 and 1001 are busy then it should go to voice mail. I have referred the following link for multi-line rollover and had used the same dial plan as mentioned https://wiki.freeswitch.org/wiki/Multi-line_rollover But when I dial 1000 from 1002 then it goes on the voice mail and the call in not getting connected even though 1000 and 1001 are not busy. Please let me know what I am missing. Thanks Sachin -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-Call-Forwarding-to-new… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Event_socket down
by k.presler＠megafit.su 25 Jul '16

25 Jul '16

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own Debian 32bit repo
by Peter 25 Jul '16

25 Jul '16

I am still facing difficulties with my build for 32 bit Intel architecture. I published the repo on my vps. I get an error when doing an apt-get update : || Get:3 http://xxxxx.com jessie/main Translation-en_US [1,893 B] Fetched 10.0 kB in 15s (660 B/s) W: Failed to fetch gzip:/var/lib/apt/lists/partial/xxxxx.com_debian_freeswitch-unstable-i386_dists_jessie_main_i18n_Translation-en%5fUS.gz Invalid file format I have been using Stanislav Sinyagin's script as a guideline. Because of a memory issue on the host machine with 512MiB RAM, I had to modify the invoked debian/rules script by Travis Cross, when building the sound packages I had to change the compression level in line 90 xz -6ev $(pkg)_$(ver).orig.tar compression level was -9e resulting in allocating 674 MiB for the operation. It doesn't make sense to me if this would be the cause of the invalid file format error as mentioned above. Someone any idea what went wrong?

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How to distinguish between normal originated and redirected by SIP 302 channels?
by Dmitry Sytchev 25 Jul '16

25 Jul '16

Hi all, We build FS-based PBX and faced issue with identifying outbound channels created by FS in case of receiving 302. We don't use internal redirect as it leads to redirecting source channel. Instead, we use enterprise originate and make FS send all channels to SBC which sends them back to FS in case of redirect. Problem is that after FS sends call redirected by 302 to SBC, we can't distinguish between usual originated calls from FS and redirect attempts as there is no significant difference in SIP fields. It would be great if we can for example set some header to identify redirected call from FS on SBC. Maybe you guys have some thoughts about how this can be done? -- Best regards, Dmitry Sytchev, IT Engineer

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mod_dp_tools att_xfer
by Luis Azedo 25 Jul '16

25 Jul '16

Hi, is there a way to make att_xfr app to hunt for dialplan without using loopback ? intended use is... <action application="att_xfer" data="2137 XML my_context"/> which doesn't work <action application="att_xfer" data="loopback/2137/my_context"/> does work but.., i heard that "loopback is evil" and you shouldn't be using it Thanks

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[ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [database or disk is full]
by Sean Devoy 23 Jul '16

23 Jul '16

HI All, Needless to say we are a non-Linux shop. Today we got FreeSwitch phone problems and the log says "[ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [database or disk is full]". Obviously something is out of space. How do we determine WHAT is out of space? I do periodically delete old FS log files. Other than that, this system DOES NOT EVER get changed. The only information I have found on disk space usage is to use "df", which seems to show 100% usage regardless, I have removed several 100 MB of files with no change. We do not use CDR data for our billing. How do I empty cdr_data and sofia reg DBs? We are offering paypal reward to clean up and leave us a "maintenance plan" commensurate with effort involved. df shows: Filesystem 1K-blocks Used Available Use% Mounted on /dev/mapper/vg_root-lv_root 145725896 144108556 0 100% / tmpfs 1029260 0 1029260 0% /dev/shm ls -l /usr/local/freeswitch/db shows: total 210704 -rw-r--r-- 1 root root 38912 Jul 22 13:57 call_limit.db -rw-r--r-- 1 root root 56908800 Jul 22 13:53 cdr_data.db -rw-r--r-- 1 root root 475136 Jul 22 13:58 core.db -rw-r--r-- 1 root root 5120 Jul 22 13:33 fifo.db -rw-r--r-- 1 root root 157506560 Jul 22 13:58 sofia_reg_external.db -rw-r--r-- 1 root root 189440 Jul 22 13:57 sofia_reg_external_noauth.db -rw-r--r-- 1 root root 362496 Jul 22 11:47 voicemail_default.db Thanks in advance. Sean

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Inbound Call Not Working
by Murugan Pandian 23 Jul '16

23 Jul '16

Hi, When i receive inbound call from gateway i am getting following log in my console but my dialplan never work. 2016-07-23 08:39:01.839304 [WARNING] switch_core_state_machine.c:646 2052065a-b589-4bed-9bde-cb42d1de7b96 sofia/internal/+17329933214(a)fl.gg Abandoned 2016-07-23 08:39:01.839304 [NOTICE] switch_core_state_machine.c:649 Hangup sofia/internal/+17329933214(a)fl.gg [CS_NEW] [WRONG_CALL_STATE] 2016-07-23 08:39:01.839304 [NOTICE] switch_core_session.c:1664 Session 23 (sofia/internal/+17329933214(a)fl.gg) Ended 2016-07-23 08:39:01.839304 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/+17329933214(a)fl.gg [CS_DESTROY] 2016-07-23 08:39:02.919323 [WARNING] switch_core_state_machine.c:646 44a357e1-c4ad-44e7-9bb5-97e7e9ba6e4b sofia/internal/+17329933214(a)fl.gg Abandoned 2016-07-23 08:39:02.919323 [NOTICE] switch_core_state_machine.c:649 Hangup sofia/internal/+17329933214(a)fl.gg [CS_NEW] [WRONG_CALL_STATE] 2016-07-23 08:39:02.919323 [NOTICE] switch_core_session.c:1664 Session 24 (sofia/internal/+17329933214(a)fl.gg) Ended 2016-07-23 08:39:02.919323 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/+17329933214(a)fl.gg [CS_DESTROY] 2016-07-23 08:39:04.119302 [WARNING] switch_core_state_machine.c:646 f772fbf0-4fda-48ec-925d-26b805452a47 sofia/internal/+17329933214(a)fl.gg Abandoned 2016-07-23 08:39:04.119302 [NOTICE] switch_core_state_machine.c:649 Hangup sofia/internal/+17329933214(a)fl.gg [CS_NEW] [WRONG_CALL_STATE] 2016-07-23 08:39:04.119302 [NOTICE] switch_core_session.c:1664 Session 25 (sofia/internal/+17329933214(a)fl.gg) Ended Regard's Murugan Pandian

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