- FreeSWITCH-users - FreeSWITCH Mailing Lists

Issue with a=fmtp:18 annexb=no
by Sam 14 Feb '14

14 Feb '14

Hi, We are facing an issue with a=fmtp:18 annexb=no. Here is the call flow, Cisco CUCM -> Cisco Cube -> FS Cisco is expecting in 200 OK , m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 * a=fmtp:18 annexb=no* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 But fs is generating without *a=fmtp:18 annexb=no* m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Because of that calls are getting fail. I am not getting where should we change setting to include that line. Log, http://pastebin.com/LDHEmCNv Please help me to resolve that issue. Thanks in advance. ----- Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-a-fmtp-18-annexb-n… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 5

Outbound displays prefix
by George Assaad 14 Feb '14

14 Feb '14

Hello, Here is a typical Outbound dial plan that we use. condition destination_number ^(\d{10})$ action set sip_h_X-accountcode=${accountcode} action set sip_h_X-Tag= action set call_direction=outbound action set hangup_after_bridge=true action set effective_caller_id_name=${outbound_caller_id_name} action set effective_caller_id_number=${outbound_caller_id_number} action set inherit_codec=true action bridge sofia/gateway/domain.com-GWname/99900$1 Things work perfect however if our client is using one of the VOIP desk phone such as Polycom IP33x or Grandstream GXP21xx, the display will show the number dialed but with the prefix which the client has no clue how of it is it. The problem is magnified when he is trying to go through the call history, the Polycom will show the dialed number with the prefix which will prevent him from dial back. The grand stream is not as bad as if you go to the call hisptry it will show what was dialed only. Is there a way to not show the prefix in the phone's display although it goes out with each number dialed. Thanks, George

1 1

Originate (Outbound) Calls: start dailplan on answer
by Grant Bagdasarian 14 Feb '14

14 Feb '14

Hello, I'm using the originate command to initiate an outbound call, which works great. The phone rings. But if the user decides to pick up after 20 seconds, the dialplan has already advanced 20 seconds into the dailplan. Is there a way to start execution of the dialplan once the call is answered? Regards, Grant

2 1

Error on FreeSwitch database
by Stefano 14 Feb '14

14 Feb '14

Hello all. After a power failure, my freeswitch put some output errors: 2014-02-14 08:05:11.105388 [ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [database is locked] BEGIN EXCLUSIVE 2014-02-14 08:05:11.105388 [CRIT] switch_core_sqldb.c:1719 ERROR [database is locked] 2014-02-14 08:05:11.105388 [ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [cannot commit - no transaction is active] COMMIT 2014-02-14 08:05:40.145389 [ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [database disk image is malformed] delete from registrations where reg_user='5136000620' and realm='shopping' and hostname='srv-014' 2014-02-14 08:05:40.165353 [ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [database disk image is malformed] delete from registrations where reg_user='5136000661' and realm='shopping' and hostname='srv-014' 2014-02-14 08:05:40.165353 [ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [database disk image is malformed] delete from registrations where expires > 0 and expires <= 1392372295 and hostname='srv-014' 2014-02-14 08:05:40.385344 [ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [database disk image is malformed] delete from registrations where reg_user='5136000626' and realm='shopping' and hostname='srv-014' 2014-02-14 08:05:40.585366 [ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [database disk image is malformed] delete from registrations where reg_user='5136000662' and realm='shopping' and hostname='srv-014' How I can fix it? Thanks Att, Assinatura - Stefano I360

2 1

execute_on_preanswer or execute_on_pre_answer?
by Fabio Mierlo 14 Feb '14

14 Feb '14

Hello, I noticed that execute_on_pre_answer variable is being mentioned as execute_on_preanswer in the whole wiki. Regards, Fabio Mierlo

1 0

Re: [Freeswitch-users] Freeswitch not putting external IP in SDP and From Header when dialing out SIPURI by external profile
by Andrew Paul 13 Feb '14

13 Feb '14

Hai brian, I given my router ip in ext-sip-ip and ext-rtp-ip. Ext-rtp-ip = router-ip Issue is fs not putting profile->extrtpip because its fail in nat checking. Regards Andrew On Sunday, February 9, 2014, Brian West <brian(a)freeswitch.org> wrote: > What values do you have in ext-sip-ip and ext-rtp-ip? > -- > Brian West > brian(a)freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > On Feb 9, 2014, at 9:09 AM, Andrew Paul <andrew.paul85(a)gmail.com> wrote: > >> Hi all, >> >> >> ------------------FREESWITCH--------------(bridge=sofia/external/sip:outbound-opensip.com:5060 )-----------OPENSIP---------------------- >> >> I am facing NAT problem when i am dialing SIP URI via sip profile. When i am dialing the SIP uri with Gateway trunk NAT is working properly. >> >> Dialing SIP URI >> >> <action application="bridge" data="sofia/external/sip: outbound-opensip.com:5060" >> >> In external profile setting I made the setting for ext-rtp-ip and ext-sip-ip. In freeswitch cli "sofia status profile external" i am able to see the external rtp and sip ip correctly. >> >> But when freeswitch sending INVITE external IPs are not set in SDP and FROM Header, instead it is using local IP. When i tried to debug the problem in switch_core_media.c freeswitch failed to check the NAT settings. >> >> switch_core_media.c >> >> SWITCH_DECLARE(switch_status_t) switch_core_media_choose_port(switch_core_session_t *session, switch_media_type_t type, int force) >> >> /* Check if NAT is detected */ >> if (!zstr(smh->mparams->remote_ip) && switch_core_media_check_nat(smh, smh->mparams->remote_ip)) { >> /* Yes, map the port through switch_nat */ >> /* Code */ >> } else { >> /* No NAT traversal required, use the profile's rtp ip */ >> use_ip = smh->mparams->rtpip; >> } >> >> Here remote IP is NULL always when i am dialing URI with external profile. When I tried to make call by "sofia/gateway/trunk/sip: outbound-opensip.com:5060" i can see the remote ip as "oubound-opensip.com" and its able to external IP in SDP. >> >> Can any body tell how to solve this problem ? Is there any configuration to set relam in dialplan ? >> >> I am using freeswitch version 1.5.6. >> >> Hopping the earliest response. >> >> Regards >> >> Andrew > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.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 > http://www.freeswitch.org >

2 7

blocked 10 secs in hangup
by Javier Menendez 13 Feb '14

13 Feb '14

I have very weird behaviour when some calls hangup, freeswitch freezes for exactly 10 seconds as the log shown below. I don't see any trace of error but for 10 seconds even the audio is not delivered, I guess is some timeout related issue but I don't know where to look any clue? 2014-02-12 18:39:25.668742 [DEBUG] switch_core_state_machine.c:579 (sofia/external/spyU0lQL2Zyb210ZG0tMDAyMDNiYjQ=(a)10.10.20.6) State DESTROY going to sleep 2014-02-12 18:39:35.668734 [DEBUG] switch_core_state_machine.c:48 rtmp/default/chanspy_001-1392227350.2513709 Standard HANGUP, cause: NORMAL_CLEARING

3 2

ded channels
by Jason Holden 13 Feb '14

13 Feb '14

Hi, When reviewing my demo fs the other day saw that fs was hanging channels in ringing state etc. Was this a bug with v1.2.20 and corrected in 1.2.21? Just update to v.1.2.21 this morning.

2 1

esl, send_dtmf, too early ESL_EVENT_CHANNEL_EXECUTE_COMPLETE
by Marcin S 13 Feb '14

13 Feb '14

Hello, In my simple C application I want to send dtmfs. This is quite easy - you just call esl_execute("send_dtmf", "1234", uuid); and it works. However, ESL returns ESL_EVENT_CHANNEL_EXECUTE_COMPLETE way to fast - even before it starts dtmfs generation! How can I know, when dtmfs were really generated? So far, the only method is to issue some sleep(), but it looks like big workaround.

5 7

routing incoming SMS?
by Peter Villeneuve 13 Feb '14

13 Feb '14

Hi, I'm experimenting with GSMopen and I'm having trouble doing something which is likely pretty simple. I have enabled mod_sms and GSMopen in FS. I can send SIP simple messages between 2 registered peers OK. I can see that incoming GSM SMS messages arrive in FS as expected, but I can't seem to get them converted into SIP SIMPLE format and routed to the peer (1000). Once they arrive in FS, I can see in the log: 2014-01-30 20:00:54.840999 [INFO] mod_sms.c:336 Processing text message +4412398746->gsm01 in context default Chatplan: gsm01 parsing [default->basic p2p] continue=true Chatplan: gsm01 Regex (PASS) [basic p2p] to(gsm01) =~ /^(.*)$/ break=on-false Chatplan: gsm01 Action send() My chatplan has: <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="basic p2p" continue="true"> <condition field="to" expression="^(.*)$">  <action application="send"/> </condition> </extension> </context> </include> And in my gsmopen.conf.xml: <configuration name="gsmopen.conf" description="GSMopen Configuration"> <global_settings> <param name="debug" value="8"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="hold-music" value="$${moh_uri}"/> <param name="destination" value="1000"/> </global_settings> I believe I'm doing something wrong in the chatplan. Is there any transfer action like in the regular XML dialplan? I tried creating in the default dialplan the following, hoping that it would be similar to sofia calls and transfer the incoming SMS to peer 1000, but no dice. <include> <extension name="sms_inbound"> <condition field="destination_number" expression="^(gsm01)$"> <action application="transfer" data="1000 XML default"/> </condition> </extension> </include> So, basically, how can I route the incoming SMS to peer 1000? Thanks

5 10