December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Dialplan for FS modems - odd behaviour or am I doing it wrong?
by Alex Crow 11 Dec '12

11 Dec '12

Hi, Please see the below log snippet. I am accepting faxes from an ISDN gateway on sequential numbers and attempting to route them into distinct freeswitch modems by taking the last two digits of the called number thusly: <condition field="destination_number" expression="^12([1-2][0-9]|3[0-6])$">  <action application="set" data="sip_ignore_reinvites=true"/> <action application="set" data="absolute_codec_string='PCMA,PCMU'"/> <action application="set" data="jitterbuffer_msec=600:600:60"/> <action application="set" data="sip_jitter_buffer_plc=false"/> <action application="bridge" data="modem/$1/a"/> </condition> And then have Hylafax email them on depending on which modem they arrived on (via FaxDispatch). However as you can see from the below log snippets, it seems that FS is quite happy to change the modem device from what is actually delivered via the dialplan. Consequently a number of my faxes are now being routed to the wrong people. Is my bridge dest wrong? Should it be something like modem/$1/0 instead of "a"? Thanks Alex Dialplan: sofia/internal/anonymous(a)anonymous.invalid Action set(sip_ignore_reinvites=true) Dialplan: sofia/internal/anonymous(a)anonymous.invalid Action set(absolute_codec_string='PCMA,PCMU') Dialplan: sofia/internal/anonymous(a)anonymous.invalid Action set(jitterbuffer_msec=600:600:60) Dialplan: sofia/internal/anonymous(a)anonymous.invalid Action set(sip_jitter_buffer_plc=false) Dialplan: sofia/internal/anonymous(a)anonymous.invalid Action bridge(modem/13/a) 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/anonymous(a)anonymous.invalid) State Change CS_ROUTING -> CS_EXECUTE 2012-12-11 14:43:18.873181 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/anonymous(a)anonymous.invalid [BREAK] 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/anonymous(a)anonymous.invalid) State ROUTING going to sleep 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/anonymous(a)anonymous.invalid) Running State Change CS_EXECUTE 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/anonymous(a)anonymous.invalid) State EXECUTE 2012-12-11 14:43:18.873181 [DEBUG] mod_sofia.c:242 sofia/internal/anonymous(a)anonymous.invalid SOFIA EXECUTE 2012-12-11 14:43:18.873181 [DEBUG] switch_core_state_machine.c:209 sofia/internal/anonymous(a)anonymous.invalid Standard EXECUTE EXECUTE sofia/internal/anonymous(a)anonymous.invalid set(call_direction=local) 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 sofia/internal/anonymous(a)anonymous.invalid SET [call_direction]=[local] EXECUTE sofia/internal/anonymous(a)anonymous.invalid set(open=true) 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 sofia/internal/anonymous(a)anonymous.invalid SET [open]=[true] EXECUTE sofia/internal/anonymous(a)anonymous.invalid hash(insert/192.168.20.245-spymap/anonymous/4e5c4699-aa22-4998-b7fa-b50a52c940fb) EXECUTE sofia/internal/anonymous(a)anonymous.invalid hash(insert/192.168.20.245-last_dial/anonymous/1213) EXECUTE sofia/internal/anonymous(a)anonymous.invalid hash(insert/192.168.20.245-last_dial/global/4e5c4699-aa22-4998-b7fa-b50a52c940fb) EXECUTE sofia/internal/anonymous(a)anonymous.invalid set(RFC2822_DATE=Tue, 11 Dec 2012 14:43:18 +0000) 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 sofia/internal/anonymous(a)anonymous.invalid SET [RFC2822_DATE]=[Tue, 11 Dec 2012 14:43:18 +0000] EXECUTE sofia/internal/anonymous(a)anonymous.invalid set(sip_ignore_reinvites=true) 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 sofia/internal/anonymous(a)anonymous.invalid SET [sip_ignore_reinvites]=[true] EXECUTE sofia/internal/anonymous(a)anonymous.invalid set(absolute_codec_string='PCMA,PCMU') 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 sofia/internal/anonymous(a)anonymous.invalid SET [absolute_codec_string]=['PCMA,PCMU'] EXECUTE sofia/internal/anonymous(a)anonymous.invalid set(jitterbuffer_msec=600:600:60) 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 sofia/internal/anonymous(a)anonymous.invalid SET [jitterbuffer_msec]=[600:600:60] EXECUTE sofia/internal/anonymous(a)anonymous.invalid set(sip_jitter_buffer_plc=false) 2012-12-11 14:43:18.873181 [DEBUG] mod_dptools.c:1344 sofia/internal/anonymous(a)anonymous.invalid SET [sip_jitter_buffer_plc]=[false] EXECUTE sofia/internal/anonymous(a)anonymous.invalid bridge(modem/13/a) 2012-12-11 14:43:18.873181 [DEBUG] switch_channel.c:1089 sofia/internal/anonymous(a)anonymous.invalid EXPORTING[export_vars] [domain_name]=[192.168.20.245] to event 2012-12-11 14:43:18.873181 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:1417 Modem /dev/FS/FS10 [ONHOOK] - Changing state to ACQUIRED 2012-12-11 14:43:18.873181 [NOTICE] switch_channel.c:968 New Channel modem/10/a [a202aef1-1628-45e8-b720-9f5368aa4565] 2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:769 modem/10/a setup codec L16/8000/20 2012-12-11 14:43:18.873181 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [rtp_autoflush_during_bridge]=[false] 2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:896 (modem/10/a) State Change CS_NEW -> CS_INIT 2012-12-11 14:43:18.873181 [DEBUG] switch_core_session.c:1283 Send signal modem/10/a [BREAK] 2012-12-11 14:43:18.873181 [DEBUG] mod_spandsp_modem.c:581 modem/10/a CHANNEL KILL 2012-12-11 14:43:18.893118 [DEBUG] switch_core_state_machine.c:415 (modem/10/a) Running State Change CS_INIT 2012-12-11 14:43:18.893118 [DEBUG] switch_core_state_machine.c:454 (modem/10/a) State INIT 2012-12-11 14:43:18.893118 [DEBUG] mod_spandsp_modem.c:461 Modem /dev/FS/FS10 [ACQUIRED] - Changing state to RINGING 2012-12-11 14:43:18.893118 [DEBUG] mod_spandsp_modem.c:1131 Modem /dev/FS/FS10 [RINGING] - RNG 1 And another one: EXECUTE sofia/internal/anonymous(a)anonymous.invalid bridge(modem/23/a) 2012-12-11 13:42:20.813107 [DEBUG] switch_channel.c:1089 sofia/internal/anonymous(a)anonymous.invalid EXPORTING[export_vars] [domain_name]=[192.168.20.245] to event 2012-12-11 13:42:20.813107 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:1417 Modem /dev/FS/FS26 [ONHOOK] - Changing state to ACQUIRED 2012-12-11 13:42:20.813107 [NOTICE] switch_channel.c:968 New Channel modem/26/a [32872ae9-6f78-45b0-af56-4b18c9de7160] 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:769 modem/26/a setup codec L16/8000/20 2012-12-11 13:42:20.813107 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [rtp_autoflush_during_bridge]=[false] 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:896 (modem/26/a) State Change CS_NEW -> CS_INIT 2012-12-11 13:42:20.813107 [DEBUG] switch_core_session.c:1283 Send signal modem/26/a [BREAK] 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:581 modem/26/a CHANNEL KILL 2012-12-11 13:42:20.813107 [DEBUG] switch_core_state_machine.c:415 (modem/26/a) Running State Change CS_INIT 2012-12-11 13:42:20.813107 [DEBUG] switch_core_state_machine.c:454 (modem/26/a) State INIT 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:461 Modem /dev/FS/FS26 [ACQUIRED] - Changing state to RINGING 2012-12-11 13:42:20.813107 [DEBUG] mod_spandsp_modem.c:1131 Modem /dev/FS/FS26 [RINGING] - RNG 1 2012-12-11 13:42:20.833137 [DEBUG] mod_spandsp_modem.c:1038 Modem /dev/FS/FS26 [RINGING] - Answering 2012-12-11 13:42:20.833137 [DEBUG] mod_spandsp_modem.c:1040 Modem /dev/FS/FS26 [RINGING] - Changing state to ANSWERED

3 8

media problem afet 302
by Miha 11 Dec '12

11 Dec '12

Hi, Scenario: fs ----> SBC---> innovaphone pbx calls goes throught and media is ok (g711u). When on other side (innovaphone pbx) someone do a 302 redirect innovaphone sends a sdb request in which prepose g711u, FS trys with g711u and send back not accabtable and media does not work. In attachment I am sending jpg of wireshark. From left to right: FS, SBC, SBC MEDIA (on different ip than sbc). How to deal with this? Thanks! Miha

3 7

Wish Lists - 'tis the season
by Sean Devoy 11 Dec '12

11 Dec '12

Hey Guys, Update those wish lists. Anthony's starts at like $495! I want to show my love, but I am a poor VOIP provider and not spending $495 on my wife this season! I hooked up BF and MC but they could use a few more suggestions in the $25 to $50 range. Anybody ever setup a contribution app so users could leave a contribution TOWARD the purchase of one of the expensive items that you guys deserve. Then others could see that and add to it? I could hook that up easy enough. Keeping track of what was purchased outright on the other list might be too confusing, but separate lists should be a breeze. Just a thought. Sean

5 7

FreeSWITCH+Sangoma with dialup modems
by Sirish Masur Mohan 11 Dec '12

11 Dec '12

Hello All, I am a newbie to FreeSWITCH, and have been experimenting with it. I am currently facing a problem, which, to be honest, I am not sure if is a FreeSWITCH misconfiguration problem, or if it is totally unrelated! My setup is as follows: Serial port based device(Caller)->Dlink dialup modem(DFM562E)->PBX->Sangoma A102 E1 Link(and FreeSWITCH)->PBX->Dlink dialup modem(DFM562E)->Server(Receiver) Problem: When the caller makes the call (using the modem with a lower baud rate 1200), Receiver modem receives the RING and answers the call (using the ATA command). But for some reason, both the ends timeout waiting for modem CONNECT. Freeswitch logs show: freeswitch@clock2> 2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s1c30][1:31] Received SETUP (suId:1 suInstId:0 spInstId:5) 2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s1c30][1:31] Incoming call: Called No:[997870] Calling No:[988869] 2012-12-10 16:53:30.975599 [NOTICE] switch_channel.c:953 New Channel FreeTDM/1:30/997870 [0f80e853-5e9b-4c33-af82-b960b261af19] 2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_out.c:175 [s1c30][1:31] Sending PROCEED (suId:1 suInstId:5 spInstId:5 dchan:1 ces:0) 2012-12-10 16:53:30.975599 [INFO] mod_dialplan_xml.c:498 Processing <988869>->997870 in context default 2012-12-10 16:53:30.975599 [NOTICE] switch_channel.c:953 New Channel FreeTDM/1:1/988713 [b7eed81e-0404-4a79-b9db-17fa59add3bd] 2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[988713] Calling No:[988869] 2012-12-10 16:53:30.975599 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:6 spInstId:0 dchan:1 ces:0) 2012-12-10 16:53:31.195550 [INFO] ftmod_sangoma_isdn_stack_rcv.c:194 [s1c1][1:1] Received PROCEED (suId:1 suInstId:6 spInstId:6 ces:0) 2012-12-10 16:53:32.695541 [INFO] ftmod_sangoma_isdn_stack_rcv.c:194 [s1c1][1:1] Received ALERT (suId:1 suInstId:6 spInstId:6 ces:0) 2012-12-10 16:53:32.695541 [NOTICE] mod_freetdm.c:2716 Ring-Ready FreeTDM/1:1/988713! 2012-12-10 16:53:32.695541 [INFO] ftmod_sangoma_isdn_stack_out.c:233 [s1c30][1:31] Sending ALERT (suId:1 suInstId:5 spInstId:5 dchan:1 ces:0) 2012-12-10 16:53:32.695541 [NOTICE] switch_ivr_originate.c:521 Ring Ready FreeTDM/1:30/997870! 2012-12-10 16:53:32.695541 [NOTICE] switch_ivr_originate.c:521 Ring-Ready FreeTDM/1:30/997870! 2012-12-10 16:53:34.635576 [INFO] ftmod_sangoma_isdn_stack_rcv.c:144 [s1c1][1:1] Received CONNECT/CONNECT ACK (suId:1 suInstId:6 spInstId:6 ces:0) 2012-12-10 16:53:34.635576 [NOTICE] mod_freetdm.c:2686 Channel [FreeTDM/1:1/988713] has been answered 2012-12-10 16:53:34.635576 [INFO] ftmod_sangoma_isdn_stack_out.c:269 [s1c30][1:31] Sending CONNECT (suId:1 suInstId:5 spInstId:5 dchan:1 ces:0) 2012-12-10 16:53:34.635576 [NOTICE] switch_ivr_originate.c:3376 Channel [FreeTDM/1:30/997870] has been answered 2012-12-10 16:53:34.735754 [INFO] ftmod_sangoma_isdn_stack_rcv.c:144 [s1c30][1:31] Received CONNECT/CONNECT ACK (suId:1 suInstId:5 spInstId:5 ces:0) And my dial plan is: <extension name="997870"> <condition field="destination_number" expression="997870"> <action application="set" data="api_hangup_hook=lua hook-test.lua"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="TEST_VAR=${lua(pool-test.lua)}"/> <action application="set" data="MY_VAR=${lua(helloworld.lua arg1 arg2)}"/> <action application="bridge" data="freetdm/wp1/a/988713"/>  </condition> </extension> Could you please help? With regards, Sirish

2 4

FreeSWITCH Weekly News and Notes
by Michael Collins 10 Dec '12

10 Dec '12

Greetings! We are glad to report that the FreeSWITCH team has tagged version 1.2.5.3. You can download the tarball here<http://files.freeswitch.org/freeswitch-1.2.5.3.tar.bz2>. Anyone using 1.2.5.x should update as soon as possible. We appreciate all those who have helped us with testing and tracking down some sneaky and pernicious little bugs. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05>we spent some time talking about the XML parser and some of its pre-processor directives. We discussed specifically how you can use the "exec" command to execute a shell script in the middle of XML processing. We also discussed a few tricks on how to look at the source code when you need to learn about some FreeSWITCH functionality that otherwise is not documented. This week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12>subject is still pending, so stay tuned! One other item I'd like to mention is that we've had several reports of FreeSWITCH success stories. We will be providing more information about those in upcoming stories on our Web site. We've got people using FreeSWITCH in various situations as well as software developers who've added support for FreeSWITCH to their offerings. The FreeSWITCH ecosystem continues to grow and flourish! Thank you all for being a part of it. Take care and have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

4 3

Hello hackers!
by Brian Foster 10 Dec '12

10 Dec '12

Regarding a recent mailing list posting that included some of my IP addresses, most of you don't know that I do set up honeypots in hopes of catching some of the bad apples that try and hack into our phone systems. We have a centralized list of Bad IP's that end up getting sent to all of our other servers. Today, one of those servers was an IT guy that works for one of my clients. He has since been fired. If anyone is interested in the 180,000 IP's I've collected...sorry you can't have 'em. -BDF Sent from my iPhone

10 18

Improve generated tones quality
by Emrah 10 Dec '12

10 Dec '12

Hi list, I am experiencing a recurring problem with generated tones across different machines and FS versions. The start or the end of the tone is often distorted. For some interesting reasons, this does not happen with all equipments and not all codecs. Typical scenario in PCMU/A: If someone calls me on an inbound DID, I pick up the call and than transfer the call to an internal extension, the original caller will hear MOH followed by a ringback that starts with a sort of "fade in". Something similar happens with the voicemail where the end of the beep tone is faded out. Even though this hasn't much to do with Early Media, I tried playing with bridge_early_media to no avail. It added a 5 seconds latency at the start of my call and did not solve my issue. Your take on this would be greatly appreciated. Is there a way to fine tune PCMU / PCMA for better performance? Can we initiate a little buffering on system generated tones? Cheers, Emrah Ps: It sounds weird, but I believe this also happens if the played tone is an actual sound file.

2 1

401
by Blake Priddy 10 Dec '12

10 Dec '12

So I updated to the latest git head and now several phones NOT ALL, including mine :( are giving me 2012-12-10 09:36:04.584347 [DEBUG] sofia_reg.c:1498 Send challenge for [ 1206(a)pbx.bryantschools.org] 2012-12-10 09:36:04.584347 [WARNING] sofia_reg.c:1502 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1206(a)pbx.bryantschools.org] from ip 10.10.24.20 send 679 bytes to udp/[10.10.24.20]:3072 at 15:36:04.585091: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.24.20:3072;branch=z9hG4bK-n757mrbdlqdw;rport=3072 From: "Blake Priddy" <sip:1206@pbx.bryantschools.org>;tag=dbou3jy9kf To: "Blake Priddy" <sip:1206@pbx.bryantschools.org>;tag=v2Kpa56439DgK Call-ID: 4e000000e08c-cm5eg6nr96al CSeq: 7373 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.3.9b+git~20121206T182231Z~a7fafb2039 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="pbx.bryantschools.org", nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 884 bytes from udp/[10.10.24.20]:3072 at 15:36:04.680840: ------------------------------------------------------------------------ REGISTER sip:pbx.bryantschools.org SIP/2.0 Via: SIP/2.0/UDP 10.10.24.20:3072;branch=z9hG4bK-cpukgis6pexk;rport From: "Blake Priddy" <sip:1206@pbx.bryantschools.org>;tag=dbou3jy9kf To: "Blake Priddy" <sip:1206@pbx.bryantschools.org> Call-ID: 4e000000e08c-cm5eg6nr96al CSeq: 7374 REGISTER Max-Forwards: 70 Contact: <sip:1206@10.10.24.20:3072 >;reg-id=1;q=1.0;+sip.instance="<urn:uuid:cb7105d7-ce2c-4835-8300-000413411CA5>" User-Agent: snom870/8.7.3.10 Allow-Events: dialog X-Real-IP: 10.10.24.20 Supported: path, gruu WWW-Contact: <http://10.10.24.20:80> WWW-Contact: <https://10.10.24.20:443> Authorization: Digest username="1206",realm="pbx.bryantschools.org ",nonce="4f0b07b6-42df-11e2-ab4a-4bb963400da6",uri="sip: pbx.bryantschools.org ",qop=auth,nc=00000001,cnonce="4afc06b9",response="8c9a1fcedd906467bd0afa2b98b50829",algorithm=MD5 Expires: 3600 Content-Length: 0 So........ Yea ha -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

4 3

IVR menu delay - 1 second between events
by Cal Leeming [Simplicity Media Ltd] 10 Dec '12

10 Dec '12

Hello, There appears to be a delay of 1 second between menu events within an IVR (i.e. play long, invalid option, play short). We have the following IVR menu; <menu name="ivr.2" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu" exit-sound="phrase:demo_ivr_main_menu" confirm-macro="" confirm-key="" confirm-attempts="3" timeout="10000" inter-digit-timeout="10" max-failures="3" max-timeouts="3" digit-len="1"> In the above example, any option will result in an invalid selection being met. However, FS seems to take its sweet time getting from the point of a user pushing a button, to actually playing the next audio. freeswitch@vded213> 2012-12-07 16:17:21.752141 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 2:2240 2012-12-07 16:17:21.752141 [DEBUG] switch_ivr_play_say.c:1682 done playing file /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav 2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:395 digits '2' 2012-12-07 16:17:21.872105 [DEBUG] switch_ivr_menu.c:579 IVR menu 'ivr.2' caught invalid input '2' 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[ivr/ivr-welcome_to_freeswitch.wav] (en:en) 2012-12-07 16:17:22.872155 [DEBUG] switch_core_file.c:180 File /usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.wav sample rate 48000 doesn't match requested rate 8000 2012-12-07 16:17:22.872155 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 20ms In the above, you can see the IVR knew the menu was invalid at "16:17:21.872105", but didn't start playing the next audio until "16:17:22.872155". The difference between the two is almost exactly 1 second, which makes me think maybe there's a config option somewhere that isn't documented/obvious, or a hard coded delay. Any advice on this would be much appreciated, slow/laggy menus are a pet hate of mine :/ Thanks Cal

3 6

sip registration
by Archana Venugopan 10 Dec '12

10 Dec '12

Hi, Currently we register authentication name as say '100' in sip registration, this comes to freeswitch and it will check in our DB for 100 and if its present then registrations would be successful. freeswitch@internal> show registrations reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname 100,fsfailover.uk01.com,e4969067f9a8c098,sofia/internal/sip:100@192.168.2.234:5060;transport=udp,1354638871,192.168.2.234,5060,udp,squay-laptop-1.squay.com I want to change this 100 to some e-mail address, so instead of 100 it will be something like 'ana(a)gmail.com'. Can we do this? While coming to freeswitch whether there would be any issues? Regards, Archana

3 12