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Getting Error message " [ERR] switch_cpp.cpp:48 Cannot queue any more events....."
by Frankie Yiu 18 May '11

18 May '11

Hi there, When I made 200 calls to the other freeswitch box in different network, I got a lot of the following error messages: [ERR] switch_cpp.cpp:48 Cannot queue any more events..... FYI--I have a 1 second delay for every 25 calls. What did I do wrong and how can I solve this? Because of these errors, I did not get any Hangup completion event. Thanks, Frankie

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skype child accounts and FS
by Madovsky 18 May '11

18 May '11

Is it possible to register multiple skype child accounts from http://www.skype.com/intl/en/business/ with FS and skype_open ? Thanks

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vm_list sends Notify to phone
by Bill Dunn - VCI Internet Services 18 May '11

18 May '11

I am using the API. When I issue a "vm_list phone#@domain" command Freeswitch sends a Notify packet to the phone. My Polycom phones give me a audible alert each time they receive one of these. Is there a way to stop Freeswitch from sending a Notify packet to the phone each time a "vm_list" command is sent to the API? Bill Dunn

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freeSWITCH segfault on openwrt
by Woody Dickson 18 May '11

18 May '11

Hi, I manage to get freeSWITCH to compile and run on openwrt, but then freeSWITCH keeps crashing when the number of calls reaches 25 consistently. Content of the core file returns nothing when opened with gdb. Does anyone have any clue as to why this is happening and how to fix it? thanks, woody

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AC Problem?
by Bill Dunn - VCI Internet Services 18 May '11

18 May '11

I have Freeswitch 1.0 running. I had been using it successfully until I took a VOIP phone home and couldn't get it to register. I tried different VOIP accounts on the box but they all do the same thing - 401 Unauthorized. I suspected it may be ACL related but not sure. I have another VOIP phone at another employee's house and it works properly without any modification to the ACLs. How can I determine if it is an ACL problem or something else? I don't get enough debugging output at the cli with "/log 7" and I've tried watching the parkcets with wireshark. Neither give me enough info to determine if the problem is an ACL or something else. Bill Dunn

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mod_lcr
by Chris Graham 18 May '11

18 May '11

Hi All, I am having an issue with mod_lcr, when I query from the CLI it returns the expected values as per: freeswitch@internal> lcr 0112341234 peak | Digit Match | Carrier | Rate | Codec | CID Regexp | Limit | Dialstring | | 011 | Telkom | 0.32000 | | | | [lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234 | freeswitch@internal> 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:1687 data passed to lcr is [0112341234 peak] 2011-05-16 17:06:19.847934 [WARNING] mod_lcr.c:1731 Using default CID [18005551212] 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:836 Has NPA NXX: [1 == 1] 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:771 0112341234 doesn't appear to be a NANP number 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:852 intra routing [state:0 lata:0] so rate field is [rate] 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:872 we have an event 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:896 SQL: SELECT l.digits AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix, cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN (0112341234, 011234123, 01123412, 0112341, 011234, 01123, 0112, 011, 01, 0) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC, rate, quality DESC, reliability DESC, rand(); 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring [lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:664 Adding Telkom to head of list 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring [lcr_carrier=Neotel,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:682 Ignoring Duplicate route for termination point (sofia/gateway/pp-ast-trunk-01/:) 2011-05-16 17:06:19.847934 [DEBUG] mod_lcr.c:337 Returning Dialstring [lcr_carrier=Telkom,lcr_rate=0.32000]sofia/gateway/pp-ast-trunk-01/0112341234 When calling from the dialplan, it fails with: 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1563 intrastate channel var is [undef] 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1566 Select routes based on interstate rates 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:1585 LCR Lookup on $1 using profile peak 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:836 Has NPA NXX: [1 == 1] 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:771 $1 doesn't appear to be a NANP number 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:852 intra routing [state:0 lata:0] so rate field is [rate] 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:857 we have a session 2011-05-16 17:08:33.100261 [DEBUG] mod_lcr.c:896 SQL: SELECT l.digits AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix, cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN (1) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC, rate, quality DESC, reliability DESC, rand(); EXECUTE sofia/internal/1000(a)192.168.16.136 bridge() 2011-05-16 17:08:33.100261 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1000(a)192.168.16.136 has executed the last dialplan instruction, hanging up. The xml entry called: <extension name="Test"> <condition field="destination_number" expression="^0112341234$"> <action application="lcr" data="$1 peak"/> <action application="bridge" data="${lcr_auto_route}"/> </condition> </extension> My custom query: <profile name="peak"> <param name="custom_sql" value=" SELECT l.digits AS lcr_digits, c.carrier_name AS lcr_carrier_name, l.rate AS lcr_rate_field, cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix, l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip, l.prefix AS lcr_prefix, l.suffix AS lcr_suffix, cg.codec AS lcr_codec, l.cid AS lcr_cid FROM lcr l JOIN carriers c ON l.carrier_id=c.id JOIN carrier_gateway cg ON c.id=cg.carrier_id WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1' AND digits IN (${lcr_query_expanded_digits}) AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end ORDER BY digits DESC, rate, quality DESC, reliability DESC, rand(); "/> </profile> Thanks in advance, Chris

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Clustering/ storing registration server IP in DB
by Anton VG 18 May '11

18 May '11

Trying to figure how to get info where (on which FS server) the SIP endpoint have been registered. There is an XML_CURL - but it get's only registration requests, and than not notified by the FS server, whether endpoint have been accepted for registration by the server or not. Should I listen for an proper event, and than update the DB ? Or there is a simplier/proper way to update DB upon endpoint registration? 2. Also - if there is a multiple regs allowed, and the same user have been registered on several FS server?

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Re: [Freeswitch-users] How to resolve AUDIO RTP REPORTS ERROR: [Bind Error!]
by Frankie Yiu 18 May '11

18 May '11

Gabe, Here are my conf files ( I have included sofia, external, and internal) ##### *sofia.conf* ** ** -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/autoload_configs/sofia.conf.xml#> <configuration name="*sofia.conf*" description="*sofia Endpoint*"> -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/autoload_configs/sofia.conf.xml#> <global_settings> <param name="*log-level*" value="*0*" /> -  <param name="*debug-presence*" value="*0*" /> </global_settings> -  -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/autoload_configs/sofia.conf.xml#> <profiles> <X-PRE-PROCESS cmd="*include*" data="*../sip_profiles/*.xml*" /> </profiles> </configuration> *##### External* -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#> <profile name="*external*"> -  -  -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#> <gateways> <X-PRE-PROCESS cmd="*include*" data="*external/*.xml*" /> </gateways> -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#> <aliases> -  </aliases> -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#> <domains> <domain name="*all*" alias="*false*" parse="*true*" /> </domains> -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external.xml#> <settings> <param name="*debug*" value="*0*" /> -  -  <param name="*sip-trace*" value="*no*" /> <param name="*rfc2833-pt*" value="*101*" /> <param name="*sip-port*" value="*$${external_sip_port}*" /> <param name="*dialplan*" value="*XML*" /> <param name="*context*" value="*public*" /> <param name="*dtmf-duration*" value="*2000*" /> <param name="*inbound-codec-prefs*" value="*$${global_codec_prefs}*" /> <param name="*outbound-codec-prefs*" value="*$${outbound_codec_prefs}*"/> <param name="*hold-music*" value="*$${hold_music}*" /> <param name="*rtp-timer-name*" value="*soft*" /> -  -  <param name="*local-network-acl*" value="*localnet.auto*" /> <param name="*manage-presence*" value="*false*" /> -  -  -  -  -  -  -  -  -  <param name="*inbound-codec-negotiation*" value="*generous*" /> <param name="*nonce-ttl*" value="*60*" /> <param name="*auth-calls*" value="*false*" /> -  <param name="*rtp-ip*" value="*$${local_ip_v4}*" /> <param name="*sip-ip*" value="*$${local_ip_v4}*" /> <param name="*ext-rtp-ip*" value="*$${external_rtp_ip}*" /> <param name="*ext-sip-ip*" value="*$${external_sip_ip}*" /> <param name="*rtp-timeout-sec*" value="*300*" /> <param name="*rtp-hold-timeout-sec*" value="*1800*" /> -  -  <param name="*tls*" value="*$${external_ssl_enable}*" /> -  <param name="*tls-bind-params*" value="*transport=tls*" /> -  <param name="*tls-sip-port*" value="*$${external_tls_port}*" /> -  <param name="*tls-cert-dir*" value="*$${external_ssl_dir}*" /> -  <param name="*tls-version*" value="*$${sip_tls_version}*" /> </settings> </profile> *###### Another file inside External folder* -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/external/external_sip.xml#> <include> <param name="*expire-seconds*" value="*60*" /> -  <param name="*register*" value="*true*" /> -  <param name="*register-transport*" value="*tcp*" /> -  <param name="*retry-seconds*" value="*30*" /> -  -  -  <param name="*contact-params*" value="*tport=tcp*" /> -  <param name="*ping*" value="*25*" /> -  </include> *##### Internal* -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#> <profile name="*internal*"> -  -  -  -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#> <aliases> -  </aliases> -  -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#> <gateways> <X-PRE-PROCESS cmd="*include*" data="*internal/*.xml*" /> </gateways> -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#> <domains> -  -  -  -  <domain name="*all*" alias="*true*" parse="*false*" /> </domains> -<file:///C:/daphne/Source/BASIPServices/freeswitch/Debug/conf/sip_profiles/internal.xml#> <settings> -  -  -  -  -  <param name="*debug*" value="*0*" /> -  -  <param name="*sip-trace*" value="*no*" /> -  <param name="*watchdog-enabled*" value="*no*" /> <param name="*watchdog-step-timeout*" value="*30000*" /> <param name="*watchdog-event-timeout*" value="*30000*" /> <param name="*log-auth-failures*" value="*true*" /> <param name="*forward-unsolicited-mwi-notify*" value="*false*" /> <param name="*context*" value="*public*" /> <param name="*rfc2833-pt*" value="*101*" /> -  <param name="*sip-port*" value="*$${internal_sip_port}*" /> <param name="*dialplan*" value="*XML*" /> <param name="*dtmf-duration*" value="*2000*" /> <param name="*inbound-codec-prefs*" value="*$${global_codec_prefs}*" /> <param name="*outbound-codec-prefs*" value="*$${global_codec_prefs}*" /> <param name="*rtp-timer-name*" value="*soft*" /> -  <param name="*rtp-ip*" value="*$${local_ip_v4}*" /> -  <param name="*sip-ip*" value="*$${local_ip_v4}*" /> <param name="*hold-music*" value="*$${hold_music}*" /> <param name="*apply-nat-acl*" value="*nat.auto*" /> -  -  -  -  -  -  -  -  -  -  -  -  <param name="*apply-inbound-acl*" value="*domains*" /> -  <param name="*local-network-acl*" value="*localnet.auto*" /> -  -  -  -  -  -  -  -  -  -  -  -  -  <param name="*record-path*" value="*$${recordings_dir}*" /> <param name="*record-template*" value="* ${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav*" /> -  <param name="*manage-presence*" value="*true*" /> -  -  -  -  -  -  <param name="*presence-hosts*" value="*$${domain},$${local_ip_v4}*" /> -  -  -  -  -  -  -  -  -  -  <param name="*inbound-codec-negotiation*" value="*generous*" /> -  -  -  -  <param name="*tls*" value="*$${internal_ssl_enable}*" /> -  <param name="*tls-bind-params*" value="*transport=tls*" /> -  <param name="*tls-sip-port*" value="*$${internal_tls_port}*" /> -  <param name="*tls-cert-dir*" value="*$${internal_ssl_dir}*" /> -  <param name="*tls-version*" value="*$${sip_tls_version}*" /> -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  <param name="*nonce-ttl*" value="*60*" /> -  -  -  -  -  -  -  -  -  -  -  -  <param name="*auth-calls*" value="*$${internal_auth_calls}*" /> -  <param name="*inbound-reg-force-matching-username*" value="*true*" /> -  <param name="*auth-all-packets*" value="*false*" /> -  <param name="*ext-rtp-ip*" value="*auto-nat*" /> <param name="*ext-sip-ip*" value="*auto-nat*" /> -  <param name="*rtp-timeout-sec*" value="*300*" /> <param name="*rtp-hold-timeout-sec*" value="*1800*" /> -  -  -  -  -  -  -  <param name="*force-register-domain*" value="*$${domain}*" /> -  <param name="*force-subscription-domain*" value="*$${domain}*" /> -  <param name="*force-register-db-domain*" value="*$${domain}*" /> -  -  -  -  -  -  -  -  -  -  -  -  -  -  <param name="*challenge-realm*" value="*auto_from*" /> -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  -  </settings> </profile> Thanks, Frankie > > ---------- Forwarded message ---------- > From: Gabriel Gunderson <gabe(a)gundy.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Tue, 17 May 2011 23:56:08 -0600 > Subject: Re: [Freeswitch-users] How to resolve AUDIO RTP REPORTS ERROR: > [Bind Error!] > On Tue, May 17, 2011 at 3:03 PM, Frankie Yiu <frankie.k.yiu(a)gmail.com> > wrote: > > 2011-05-17 12:40:36.061473 [ERR] sofia_glue.c:3449 AUDIO RTP REPORTS > ERROR: > > [Bind Error!] > > Sounds like the IP/port is in use already. Can you pastebin your sofia > conf? > > Best, > Gabe > > >

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how to handling REFER from remote subscriber on the FreeSWICH?
by Иван Чистяков 18 May '11

18 May '11

how to handling REFER from remote subscriber on the FreeSWICH?

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ESL Bridge
by Antonio Teixeira 18 May '11

18 May '11

Good Morning. Here i continue my fight against ESL :) On Leg A I'm forking a call that will become Leg B after a short IVR. Anyway Leg A is valet_parked with MOH while i Dial Leg B , Leg B Answers , Presses 1 and is merged by uuid_bridge Then... Both calls are disconnected with Executed Last Dialplan Command . I have tried everything from freezing the IVR with sleep(in python and using the freeswitch sleep). But i'm unable to merge both calls. As anyone tried something similar. Regards A/T

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