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Blind Transfer
by Scott 09 Feb '12

09 Feb '12

Hi list, Trying to resolve an issue with blind transfers. When transferring a call, the A leg is terminated if a blind transfer is started, an attended transfer doesn't create this scenario. Here are the transcript logs from FS relating to the event. I have log level set to 6 but can increase this if I need more information. Call flow is as follows.. ext18: Calls ext51 ext18: transfers call to ext15 ext15: rings for a split second ext51: lands in voicemail of 15 ext18: clears as normal OS: CentOS release 5.7 (Final) freeswitch@internal> version FreeSWITCH Version 1.0.head (git-9de1e1a 2012-02-07 14-36-22 -0500) +OK log level [7] freeswitch@internal> 2012-02-08 16:08:10.291633 [NOTICE] switch_channel.c:920 New Channel sofia/internal/18(a)10.100.0.9 [6fd7a0b0-c518-42ff-bb24-e2acf8121b9e] 2012-02-08 16:08:10.291633 [INFO] mod_dialplan_xml.c:481 Processing Scott <18>->51 in context default 2012-02-08 16:08:10.311819 [NOTICE] switch_channel.c:920 New Channel sofia/internal/sip:51@10.100.0.16:1024 [f8f27c7d-9198-4225-a94d-2eadb9bb58a8] 2012-02-08 16:08:10.391998 [NOTICE] sofia.c:5458 Ring-Ready sofia/internal/sip:51@10.100.0.16:1024! 2012-02-08 16:08:10.391998 [INFO] switch_ivr_originate.c:1115 Sending early media 2012-02-08 16:08:10.411314 [NOTICE] sofia_glue.c:3899 Pre-Answer sofia/internal/18(a)10.100.0.9! 2012-02-08 16:08:10.431704 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0xb6ed5160 0x8b23a08 2012-02-08 16:08:10.731162 [INFO] mod_com_g729.c:148 DECODER CREATE - 0xb6ed5108 0x8c094c0 2012-02-08 16:08:11.977555 [NOTICE] sofia.c:6070 Channel [sofia/internal/sip:51@10.100.0.16:1024] has been answered 2012-02-08 16:08:11.996869 [NOTICE] switch_ivr_originate.c:3209 Channel [sofia/internal/18(a)10.100.0.9] has been answered 2012-02-08 16:08:12.098595 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xb4d025b0 (nil) 2012-02-08 16:08:12.098595 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xb4d025b0 (nil) 2012-02-08 16:08:12.177926 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xb4d02558 (nil) 2012-02-08 16:08:12.177926 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xb4d02558 (nil) 2012-02-08 16:08:14.183102 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0xb4d029e0 0xb6e6ac00 2012-02-08 16:08:15.750004 [NOTICE] switch_ivr.c:1711 Transfer sofia/internal/sip:51@10.100.0.16:1024 to XML[15@default] 2012-02-08 16:08:15.769362 [NOTICE] switch_ivr_bridge.c:1372 Hangup sofia/internal/18(a)10.100.0.9 [CS_EXECUTE] [NORMAL_CLEARING] 2012-02-08 16:08:15.789438 [INFO] mod_dialplan_xml.c:481 Processing Scott <18>->15 in context default 2012-02-08 16:08:15.789438 [INFO] switch_channel.c:2695 sofia/internal/sip:51@10.100.0.16:1024 Flipping CID from "Scott" <18> to "Outbound Call" <51> 2012-02-08 16:08:15.789438 [NOTICE] switch_channel.c:920 New Channel sofia/internal/sip:15@10.100.0.15:5064 [acf45601-79ad-4518-907f-7a58155748ab] 2012-02-08 16:08:16.231456 [NOTICE] sofia.c:5458 Ring-Ready sofia/internal/sip:15@10.100.0.15:5064! 2012-02-08 16:08:16.291519 [INFO] mod_com_g729.c:148 DECODER CREATE - 0xb4d029f8 0xb6c41798 2012-02-08 16:08:16.291519 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/sip:15@10.100.0.15:5064 [CS_CONSUME_MEDIA] [NO_ANSWER] 2012-02-08 16:08:16.291519 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [NO_ANSWER] 2012-02-08 16:08:16.291519 [INFO] mod_dptools.c:2884 Originate Failed. Cause: NO_ANSWER 2012-02-08 16:08:16.291519 [NOTICE] switch_core_session.c:1395 Session 991 (sofia/internal/sip:15@10.100.0.15:5064) Ended 2012-02-08 16:08:16.291519 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/sip:15@10.100.0.15:5064 [CS_DESTROY] 2012-02-08 16:08:16.371076 [NOTICE] switch_core_session.c:1395 Session 989 (sofia/internal/18(a)10.100.0.9) Ended 2012-02-08 16:08:16.371076 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/18(a)10.100.0.9 [CS_DESTROY] 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xb6ed5108 (nil) 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xb6ed5108 0x8c094c0 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0xb6ed5108 0x8c094c0 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xb6ed5160 0x8b23a08 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xb6ed5160 (nil) 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0xb6ed5160 0x8b23a08 2012-02-08 16:08:17.292845 [NOTICE] switch_channel.c:920 New Channel loopback/app=voicemail:default 10.100.0.9 15-a [7b76ae61-3465-40bb-9d65-41f6cba2c3d2] 2012-02-08 16:08:17.292845 [NOTICE] switch_channel.c:918 Rename Channel loopback/app=voicemail:default 10.100.0.9 15-a->loopback/voicemail-a [7b76ae61-3465-40bb-9d65-41f6cba2c3d2] 2012-02-08 16:08:17.292845 [NOTICE] switch_channel.c:920 New Channel loopback/voicemail-b [e02cc153-8651-422c-bb98-f6f7a3244418] 2012-02-08 16:08:17.292845 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/voicemail-a! 2012-02-08 16:08:17.292845 [NOTICE] mod_dptools.c:1143 Pre-Answer loopback/voicemail-b! 2012-02-08 16:08:17.414279 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x8b6e120 0xb6ec4118

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Can someone help me with a complicated configuration?
by Josue Diaz Cruz 09 Feb '12

09 Feb '12

I have a server with 4 ETH. Each one have a different IP In 2 cases i fixed the ip address, In the other 2 cases i have 2 cable connections in bridge mode so i receive the public ip in dhcp mode. eth0 local address 192.168.1.100 eth1 local address 192.168.2.100 eth2 public address dhcp (cable) eth3 public address dhcp (cable) I was thinking to build 4 different profiles for each eth port, but i came to know that if we change the place of colocation it will not work properly because the dhcp ip. It is any way to attach a profile to the eth port, not the ip? Or if you know a better way to manage this case. I would like to receive traffic from the 2 public ip's and route it both like if i have just one profile. Josue Diaz Cruz Departamento Tecnico y Soporte <mailto:jdiaz@coinfru.com> jdiaz(a)coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia <http://www.coinfru.com/> www.coinfru.com

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New to FS
by chris 09 Feb '12

09 Feb '12

Hello, I am new to FS but I'm impressed already with my first experiences with it. I was just wondering about somethins that I've always wanted on asterisk that was possible but always been a pain. Is it possible to do source based routing and route certain SIP extensions/devices to certain outbound trunks? If so how can I do it easily? Thanks for any advice / info, chris

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T.38 SDP not forwarded by FS in t38-passthru mode
by Stephen Wilde 09 Feb '12

09 Feb '12

I have trouble with fax because the INVITE generated by FS doesn't contain the image part. In my profile I have set: <param name="t38-passthru" value="true" /> and the dialplan (simplified) is: <action application="export" data="t38_passthru=true"/> <action application="bridge" data="sofia/external/xxxx" /> but when is received an INVITE with SDP: v=0 o=- 2598299616 0 IN IP4 213.204.31.103 s=session t=0 0 m=audio 4640 RTP/AVP 18 8 96 c=IN IP4 213.204.31.32 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 m=image 4642 udptl t38 c=IN IP4 213.204.31.32 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF is generated in the outbound leg an SDP: v=0 o=FreeSWITCH 1328751960 1328751961 IN IP4 217.19.146.173 s=FreeSWITCH c=IN IP4 217.19.146.173 t=0 0 m=audio 31196 RTP/AVP 18 8 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 without image part. Here the log: http://pastebin.freeswitch.org/18333 Any advice?

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302 redirect
by Miha Zoubek 09 Feb '12

09 Feb '12

Hi, I have set 302 redicet as is described on wiki (everything works ok). Problem that I am dealing with is regarding radius CDR issue. On FS I have set radius for AAA. As I do not need radius AAA for incoming calls, so I have set variable disable_radius_stop and disable_radius_start on true in public dialplan. After I set on phone 302 redirect the call should be seen in my sql (radius cdr). beacuse the variable is fist set (disable_radius_) on true I am not getting radius start and stop packet. After my dialplan recognized 302 redirect I set disable_radius_ on fals but it do not work because of the previous action (disable_radius_, I disable radius for start and stop). Please help as I do not know how to get this working. I also can not make a condition of the 302 redirected number as this variable is set further in dialplan. p.s.: in vars.xml I have set to radius diasble as I do need it only b leg. Thanks! BR, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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System shown no route but connected with false caller ID
by Man 09 Feb '12

09 Feb '12

User registered to the system and try to make call to PSTN network but system show "No Route, Aborting" at first and connect the call right after it. The destination number ring wrong caller ID displayed. 40% calls happened with this issue. Any possible reason? FreeSWITCH Version 1.0.head (git-7f5b8fb 2012-02-02 20-41-15 -0600) Checked the CDR both normal and fault call are no different. My dialplan is simply bridge the call to the voice gateway: <extension name="outbound"> <condition field="destination_number" expression="^(\d{8})$"> <action application="bridge" data="sofia/external/$1(a)203.194.x.x"/> </condition> </extension> Thanks a lot. Armand

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Extend feature for HuaWei Softswitch
by liu rush 09 Feb '12

09 Feb '12

Dear, I am currently working on a project that the situation is similar to Hybrid SIPS encryption in below link as I think. http://wiki.freeswitch.org/wiki/SIP_TLS#Hybrid_Encryption My actual situation is like this. Since, the HuaWei Softswitch isn't support for SRTP, i suppose Freeswitch can be setup as an intermediary to do it which it will be look like this. IP-phone <-SRTP-> FS <-RTP-> HuaWei Softswitch. But I have no idea how to set it up with Freeswitch (I'm newbie in FS, I jumped into FS for less than a month). Could anyone give me some advice or direction if FS is capable to do it? Regard, Rush

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Re: [Freeswitch-users] external_sip_ip not refreshed after ip address change.
by Anurag Srivastava 08 Feb '12

08 Feb '12

I have a question about nat behavior in freeswitch. Basically my external calls to freeswitch are getting disconnected after 30 seconds of two way audio when my external ip address changes. I have dhcp from my ISP and am using external_sip_ip and external_rtp_ip as stun:<stun_server>. When my IP changes I see that external_sip_ip does not get refreshed but external_rtp_ip does. I am not allowed to enable upnp/nat-pmp on my router. Apparently it is a known issue that external_sip_ip is read just at load time and not refreshed even if it is specified in stun format Is there a fix to this problem? There is always the option of restarting profile when ddclient notes an ip change. Is there something inbuilt into FS. It is already finding that ip address has changed as reflected in external_rtp_ip which does use stun and gets the right ip address.

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Polycom BLF
by DJB International 08 Feb '12

08 Feb '12

Does anyone ever experience with BLF for DND on polycom expansion module? The BLF is working fine when the extension is busy, but when the extension has DND on, it did not change the LED light. Any sugestion? Thank you.

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Re: [Freeswitch-users] Fs ignoring voicemail to email
by Balamurugan Mahendran 08 Feb '12

08 Feb '12

I did, also I have no issue sending email outside FS. Thanks, Bala On Tue, Feb 7, 2012 at 8:31 PM, Balamurugan Mahendran < balamurugan(a)adaptavant.com> wrote: > I did, also I have no issue sending email outside FS. > > Thanks, > Bala > > On Tue, Feb 7, 2012 at 8:09 PM, John <freeswitch(a)earthspike.net> wrote: > >> I updated the wiki mod_voicemail page a few days ago with some >> instructions for debugging email from freeswitch. Have you tried those? >> >> >> On 07/02/12 13:47, Balamurugan Mahendran wrote: >> >> All, >> >> I am having the same issue, its not sending email(extension --> lua >> script) getting *"Segmentation fault"* >> >> xml : >> >> <include> >> <user id="503"> >> <params> >> <param name="password" value="1234"/> >> <param name="vm-password" value="503"/> >> <param name="vm-mailto" value="b2m(a)a-cti.com"/> >> <param name="vm-enabled" value="true"/> >> <param name="vm-skip-instructions" value="true"/> >> <param name="vm-disk-quota" value="3600"/> >> <param name="vm-email-all-messages" value="true"/> >> <param name="max-login-attempts" value="2"/> >> <param name="digit-timeout" value="10000"/> >> <param name="max-record-len" value="900"/> >> <param name="tone-spec" value="%(1000, 0, 640)"/> >> <param name="record-silence-threshold" value="2000"/> >> <param name="record-silence-hits" value="3"/> >> </params> >> <variables> >> <variable name="toll_allow" value="domestic,international,local"/> >> <variable name="accountcode" value="503"/> >> <variable name="user_context" value="default"/> >> <variable name="effective_caller_id_name" value="Extension 503"/> >> <variable name="effective_caller_id_number" value="503"/> >> <variable name="outbound_caller_id_name" >> value="$${outbound_caller_name}"/> >> <variable name="outbound_caller_id_number" >> value="$${outbound_caller_id}"/> >> <variable name="callgroup" value="techsupport"/> >> </variables> >> </user> >> </include> >> >> dialplan : >> >> <include> >> <extension name="503"> >> <condition field="destination_number" expression="^(503)$"> >> <action application="lua" data="vmbox.lua >> ${effective_caller_id_number}"/> >> </condition> >> </extension> >> </include> >> >> >> Lua : >> >> caller=503; >> freeswitch.consoleLog("info","From :"..caller); >> session:set_tts_parms("flite", "slt"); >> session:speak("Welcome To Voice Mail !. You Can Leave Your Message >> Here."); >> path="/usr/local/freeswitch/recordings/"; >> prompt=caller..".mp3"; >> recpath=path..prompt; >> freeswitch.consoleLog("info","record path="..recpath); >> session:recordFile(recpath,30,10,10); >> session:speak("Thank you."); >> >> freeswitch.consoleLog("info","testing"); >> freeswitch.email("b2m(a)a-cti.com", >> "saraswathi.devaraj(a)a-cti.com", >> "subject: Voicemail from 801\n", >> "Hello,\n\nYou've got a voicemail, click the attachment >> to listen to it.", >> "/usr/local/freeswitch/recordings/503.mp3", >> "", >> ""); >> freeswitch.consoleLog("info","hai"); >> >> >> >> switchconf : >> >> >> <param name="mailer-app" value="/usr/sbin/sendmail"/> >> <param name="mailer-app-args" value="-t"/> >> >> >> Thanks for your help!! >> >> Thanks, >> Bala >> >> >> >> On Sat, Feb 4, 2012 at 12:31 AM, John <freeswitch(a)earthspike.net> wrote: >> >>> On 03/02/12 18:00, Thomas Hoellriegel wrote: >>> > >>> > Its works fine!! >>> Good news! >>> >> >> >> _________________________________________________________________________ >> 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 >> >> >

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