November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Joining a call from Gtalk to a conference on FreeSwitch
by Papineni, Suneel 22 Nov '11

22 Nov '11

Hi, I got the new GIT version and enabled mod_dingling and compiled. Everything went through and able to establish call to an extension if I configure that extension number in "client" profile. What I am trying to do is, I want to bridge or join a call coming from GTalk to an existing conference in FreeSwitch. For this purpose I configured a different number on "client profile" and created a dial-plan for this number to 'park' the call first before trying to join to the conference. Then using eventSockets I am trying to join this call to conference and issued following command. (tried with "uuid_bridge" command as well) "api uuid_transfer [Unique-ID] conference:xyz@default inline" Command is successful and also I can hear a sound that someone joined in the conference, but I didn't hear any voice at either side. I couldn't see any RTP flow as well (checked wireshark traces at FS). After sometime like 30 seconds call at GTalk is disconnected automatically. I am not sure why nothing is heard at both sides and why call got disconnected. Also tried answering the call first (after Park) and then bridging to conference, still got the same issue. Could someone please let me know if I am missing anything or need to configure in a different way for conferencing. Thanks & Regards Suneel Client.xml <profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="user(a)gmail.com/gtalk<mailto:user@gmail.com/gtalk>"/> <param name="password" value="password"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="123456"/> <param name="vad" value="both"/> </profile> Dial-plan.. <include> <extension name="GTalk_DP"> <condition field="caller_id_number" expression="^([^@]+)" break="never"> <action application="set" data="effective_caller_id_number=$1" /> </condition> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="bypass_media=true"/> <action application="park" /> </condition> </extension> </include>

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(no subject)
by Charles Wang 22 Nov '11

22 Nov '11

http://fitnessembassy.com/modules/mod_wdbanners/site.php?html143

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401 Unauthorized / phone, router or my fs config?
by Charlie Orford 22 Nov '11

22 Nov '11

Hello list I am an asterisk refugee and currently in the midst of moving our voip platform across to freeswitch. The goal is to have FS in the cloud (on a dedicated Linode virtual machine running Debian Squeeze), with all office phones (Aastra 57i units) connecting via the public internet. FS is compiled and running on the linode machine (using the latest git build from a week ago). It is setup to listen on the public IP only so there is no NAT happening at the server end. All relevant firewall ports are open (tcp/udp 5060, tcp/udp 5080 and udp 16384:32768). Because our office net connection has a dynamic IP, we are using (or trying to use) digest authentication rather than ACLs in order to control user/extension access to the internal sip profile. The problem: For some reason, none of our phones are able to successfully register with FS. Running fs_cli with logging at 7 and enabling "sofia global siptrace on" shows that the phones make contact and try to REGISTER but when FS replies with a 401 Unauthorized and requests the phone authenticate via digest, the phone seems to ignore this and just repeatedly keeps sending the same original REGISTER request with no accompanying Authorization header. My hunch is that the problem must lie with the phone or our router rather than FS but I'm a little out of my depth with this problem and so would appreciate any insight or advice. For a transcript of a failed registration between our FS server and a phone at the office, please see: http://pastebin.com/1qRudrvE (note: server and phone ip has been changed to protect the innocent). I also have a screen shot of the phone's SIP config here: http://imgur.com/2lwiN (we are running the latest publically available Aastra firmware on the phones - v3.2.2.56). Finally, in case it is relevant, the router at the office is a Draytek Vigor 2600 ADSL router (about 5 years old now but working happily as far as we know). Thanks + Regards, Charlie

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Maximum number of gateways
by Hynek Cihlar 22 Nov '11

22 Nov '11

Hello all, is there a practical limit of the number of gateways per one Sofia profile? The documentation says that Sofia is single-threaded. Does it for example mean, that if a gateway is being registered that other net IO of the same profile is stalled? Thanks, Hynek

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Re: [Freeswitch-users] Sip client java based for mobile phone
by bwibowo＠gmail.com 22 Nov '11

22 Nov '11

Not sure this can be run on old mobile phone that support java. ------Original Message------ From: EdPimentl Sender: freeswitch-users-bounces(a)lists.freeswitch.org To: FreeSWITCH Users Help ReplyTo: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sip client java based for mobile phone Sent: Nov 22, 2011 9:15 PM http://sipdroid.org/ _________________________________________________________________________ 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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Sip client java based for mobile phone
by bwibowo＠gmail.com 22 Nov '11

22 Nov '11

As asked in subject, if somebody have this kind of product, please contact me directly Regards Budi wibowo

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Fwd: Newfies-Dialer - Voice Broadcast Application Launched
by Michel Daggelinckx 22 Nov '11

22 Nov '11

---------- Forwarded message ---------- From: Star2Billing <sales(a)star2billing.com> Date: Mon, Nov 21, 2011 at 3:10 PM Subject: Newfies-Dialer - Voice Broadcast Application Launched To: "michel.daggelinckx(a)gmail.com" <michel.daggelinckx(a)gmail.com> You are receiving this mail because you have downloaded one of Star2Billing's products (A2Billing, CDR-Stats, Newfies-Dialer) or we have installed it on your behalf. If you have no further interest in any of Star2Billing's <http://star2billingsl.cmail2.com/t/r/l/juetjd/xktjkhkuj/i/>products, please unsubscribe <http://star2billingsl.cmail2.com/t/r/u/juetjd/xktjkhkuj/r/>. Having trouble reading this? View it in your browser<http://star2billingsl.cmail2.com/t/r/e/juetjd/xktjkhkuj/y/> . [image: header bg] Newfies-Dialer - Voice Broadcasting Star2Billing S.L. launches Newfies-Dialer, a new application for mass voice broadcasting [image: space] Monday 21 November [image: space] [image: space] Newfies-Dialer 1.0 Released. [image: space] [image: space] <http://star2billingsl.cmail2.com/t/r/l/juetjd/xktjkhkuj/d/> Star2Billing S.L. is proud to announce the stable release of their latest product, Newfies-Dialer (http://www.newfies-dialer.org<http://star2billingsl.cmail2.com/t/r/l/juetjd/xktjkhkuj/u/>), a free and open source voice broadcast application for Freeswitch to enable the automated delivery of interactive phone calls to contacts, clients and the general public. Voice Broadcasting is a powerful business tool to automate phone calls for appointment reminders, delivery confirmations, event promotion, debt control, IVR campaigns, phone voting and surveys, automated teleconferencing and to relay vital information to your contacts over the phone. Newfies-Dialer’s voice broadcast system is powerful and adaptable: *> Scalable* – Newfies-Dialer uses distributed message passing to make millions of calls per day via multiple cloud based instances of Freeswitch. *> Multi-User* – Facilitates SaaS voice broadcast providers to let their customers create voice broadcast campaigns via an intuitive web interface. *> Integration* – Comprehensive API ‘s to integrate Newfies-Dialer into third party applications to automate call distribution. The Newfies-Dialer Voice Broadcasting Platform is assembled entirely from free and open source components including Freeswitch, Django, Plivo, Celery and RabbitMQ. For more details on Newfies-Dialer including comprehensive installation guides, an automated install script, user manuals and developer’s documentation view www.newfies-dialer.org<http://star2billingsl.cmail2.com/t/r/l/juetjd/xktjkhkuj/o/>, or contact us at newfies-dialer(a)star2billing.com [image: space] [image: space] [image: space] Other Information Get Started<http://star2billingsl.cmail2.com/t/r/l/juetjd/xktjkhkuj/p/> Install Newfies-Dialer with automated install scripts and step-by-step instructions. Newfies-Dialer Features<http://star2billingsl.cmail2.com/t/r/l/juetjd/xktjkhkuj/x/> Features and benefits of Newfies-Dialer [image: Forward to Friend]<http://star2billingsl.forwardtomyfriend.com/r/xktjkhkuj/C7262377/juetjd/l/j> [image: Unsubscribe] <http://star2billingsl.cmail2.com/t/r/u/juetjd/xktjkhkuj/t/> [image: Follow us on Twitter]<http://star2billingsl.cmail2.com/t/r/l/juetjd/xktjkhkuj/m/> [image: header bg]

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How to disable one modul
by Miha Zoubek 22 Nov '11

22 Nov '11

Hi, I would like to disable mod_radius_cdr to start for the first time. I would like to have only the second one. Can someone help me where on configuration files or in which .c file this is set? 2011-11-22 09:16:35.541448 [DEBUG] switch_channel.c:1844 (sofia/internal/018108500(a)xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING 2011-11-22 09:16:35.541448 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/018108500(a)xxx.xxx.xxx.xxx) State ROUTING 2011-11-22 09:16:35.541448 [DEBUG] mod_sofia.c:148 sofia/internal/018108500(a)xxx.xxx.xxx.xxxSOFIA ROUTING 2011-11-22 09:16:35.541448 [DEBUG] mod_radius_cdr.c:156 [mod_radius_cdr] Entering my_on_routing, test, test 2011-11-22 09:16:35.541448 [DEBUG] mod_radius_cdr.c:374 [mod_radius_cdr] RADIUS Accounting OK 2011-11-22 09:16:35.541448 [DEBUG] switch_core_state_machine.c:104 sofia/internal/018108500(a)xxx.xxx.xxx.xxxStandard ROUTING 2011-11-22 09:16:37.551441 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:018108501@xxx.xxx.xxx.xxx:1417 [BREAK] 2011-11-22 09:16:37.551441 [DEBUG] mod_radius_cdr.c:156 [mod_radius_cdr] Entering my_on_routing, test, test 2011-11-22 09:16:37.551441 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:018108501@xxx.xxx.xxx.xxx:1417 [BREAK] 2011-11-22 09:16:37.551441 [DEBUG] mod_radius_cdr.c:374 [mod_radius_cdr] RADIUS Accounting OK 2011-11-22 09:16:37.551441 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:018108501@xxx.xxx.xxx.xxx:1417) State ROUTING Regards, Miha

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SIP invalid call attempts from unknown dialer
by D M 22 Nov '11

22 Nov '11

Hello, I have noticed quite a few different attempts on accessing my freeswitch machine via SIP that does not come from the company. I have attached part of a log with such an attempt. My main concern is ensuring that similar attempts will not be able to make external calls. I realize that any number made available externally will also be accessable via this method so my secondary concern is throttling or preventing these type of attempts to avoid autodialer spam. This log repeats with around 12 call attempts per second for almost a minute times with different attempts on seemingly random numbers. There has been multiple different attempts to spam random numbers via SIP but so far none has been successful. This log is the most relevant since it made a single login attempt on an nonexistent user after which it has either successfully spoofed the ip of the freeswitch machine or used an vulnerability in either my config or freeswitch. My config is the default freeswitch+fusionpbx installation on Ubuntu 10.04.3 LTS with instructions from here (http://wiki.fusionpbx.com/index.php?title=Easy_Ubuntu_10.04&oldid=1574) With a few minor configuration changes: * Registering is done via external domain pointing to the freeswitch machine, NOT using the default port 5060 * Port 5060 is generally used for traffic with SIP provider that connects us to phone network but the port not firewalled/restricted in any other way This is an example log of a single login attempt and single call attempt, the following modifications have been made: * Freeswitch public ip has been changed to xx.xx.xx.xx * 2 regexps have been changed from public telephone number to /^publicnumber$/ and /^publicnumber2$/ * A large list of regexps have been replaced with  Please let me know if you need any more details or longer logs Thanks, Daniel ##### LOG BEGIN ##### 2011-11-18 15:27:33.293146 [WARNING] sofia_reg.c:2283 Can't find user [1010(a)xx.xx.xx.xx] You must define a domain called 'xx.xx.xx.xx' in your directory and add a user with the id="1010" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2011-11-18 15:27:36.633145 [NOTICE] switch_channel.c:897 New Channel sofia/external/1010@xx.xx.xx.xx:5060 [75cf1808-11f1-11e1-9c95-494fea388543] 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5084 Channel sofia/external/1010@xx.xx.xx.xx:5060 entering state [received][100] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010@xx.xx.xx.xx:5060) Running State Change CS_NEW 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5095 Remote SDP: v=0^M o=1010 13216264671138 13216264671138 IN IP4 192.168.1.3^M s=VaxSoft^M c=IN IP4 192.168.1.3^M t=0 0^M m=audio 7000 RTP/AVP 0 8 3 98 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:3 GSM/8000^M a=rtpmap:98 iLBC/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:343 (sofia/external/1010@xx.xx.xx.xx:5060) State NEW 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4711 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000] 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4711 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/1010@xx.xx.xx.xx:5060 PCMA/8000 20 ms 160 samples 64000 bits 2011-11-18 15:27:36.633145 [DEBUG] sofia_glue.c:4825 Set 2833 dtmf send/recv payload to 101 2011-11-18 15:27:36.633145 [DEBUG] sofia.c:5284 (sofia/external/1010@xx.xx.xx.xx:5060) State Change CS_NEW -> CS_INIT 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010@xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010@xx.xx.xx.xx:5060) Running State Change CS_INIT 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1010@xx.xx.xx.xx:5060) State INIT 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:85 sofia/external/1010@xx.xx.xx.xx:5060 SOFIA INIT 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:125 (sofia/external/1010@xx.xx.xx.xx:5060) State Change CS_INIT -> CS_ROUTING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010@xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:364 (sofia/external/1010@xx.xx.xx.xx:5060) State INIT going to sleep 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010@xx.xx.xx.xx:5060) Running State Change CS_ROUTING 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:1821 (sofia/external/1010@xx.xx.xx.xx:5060) Callstate Change DOWN -> RINGING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:373 (sofia/external/1010@xx.xx.xx.xx:5060) State ROUTING 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:148 sofia/external/1010@xx.xx.xx.xx:5060 SOFIA ROUTING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:77 sofia/external/1010@xx.xx.xx.xx:5060 Standard ROUTING 2011-11-18 15:27:36.633145 [INFO] mod_dialplan_xml.c:336 Processing MyName <1010>->972592182076 in context public Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 parsing [public->unloop] continue=false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 parsing [public->outside_call] continue=true Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Absolute Condition [outside_call] Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Action set(outside_call=true) Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 parsing [public->call_debug] continue=true Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Regex (FAIL) [public_extensions] destination_number(972592182076) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 parsing [public->TEMP] continue=false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Regex (PASS) [TEMP] context(public) =~ /public/ break=on-false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Regex (FAIL) [TEMP] destination_number(972592182076) =~ /^publicnumber$/ break=on-false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 parsing [public->Misc_Number] continue=false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Regex (PASS) [Misc_Number] context(public) =~ /public/ break=on-false Dialplan: sofia/external/1010@xx.xx.xx.xx:5060 Regex (FAIL) [Misc_Number] destination_number(972592182076) =~ /^publicnumber2$/ break=on-false  2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:119 (sofia/external/1010@xx.xx.xx.xx:5060) State Change CS_ROUTING -> CS_EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010@xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:373 (sofia/external/1010@xx.xx.xx.xx:5060) State ROUTING going to sleep 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010@xx.xx.xx.xx:5060) Running State Change CS_EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:380 (sofia/external/1010@xx.xx.xx.xx:5060) State EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:241 sofia/external/1010@xx.xx.xx.xx:5060 SOFIA EXECUTE 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:157 sofia/external/1010@xx.xx.xx.xx:5060 Standard EXECUTE EXECUTE sofia/external/1010@xx.xx.xx.xx:5060 set(outside_call=true) 2011-11-18 15:27:36.633145 [DEBUG] mod_dptools.c:1063 sofia/external/1010@xx.xx.xx.xx:5060 SET [outside_call]=[true] EXECUTE sofia/external/1010@xx.xx.xx.xx:5060 set(RFC2822_DATE=Fri, 18 Nov 2011 15:27:36 +0100) 2011-11-18 15:27:36.633145 [DEBUG] mod_dptools.c:1063 sofia/external/1010@xx.xx.xx.xx:5060 SET [RFC2822_DATE]=[Fri, 18 Nov 2011 15:27:36 +0100] 2011-11-18 15:27:36.633145 [NOTICE] switch_core_state_machine.c:189 sofia/external/1010@xx.xx.xx.xx:5060 has executed the last dialplan instruction, hanging up. 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:2739 (sofia/external/1010@xx.xx.xx.xx:5060) Callstate Change RINGING -> HANGUP 2011-11-18 15:27:36.633145 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/external/1010@xx.xx.xx.xx:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2011-11-18 15:27:36.633145 [DEBUG] switch_channel.c:2755 Send signal sofia/external/1010@xx.xx.xx.xx:5060 [KILL] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010@xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:380 (sofia/external/1010@xx.xx.xx.xx:5060) State EXECUTE going to sleep 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010@xx.xx.xx.xx:5060) Running State Change CS_HANGUP 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:575 (sofia/external/1010@xx.xx.xx.xx:5060) State HANGUP 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:458 Channel sofia/external/1010@xx.xx.xx.xx:5060 hanging up, cause: NORMAL_CLEARING 2011-11-18 15:27:36.633145 [DEBUG] mod_sofia.c:522 Responding to INVITE with: 480 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:46 sofia/external/1010@xx.xx.xx.xx:5060 Standard HANGUP, cause: NORMAL_CLEARING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:575 (sofia/external/1010@xx.xx.xx.xx:5060) State HANGUP going to sleep 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1010@xx.xx.xx.xx:5060) State Change CS_HANGUP -> CS_REPORTING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_session.c:1154 Send signal sofia/external/1010@xx.xx.xx.xx:5060 [BREAK] 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:325 (sofia/external/1010@xx.xx.xx.xx:5060) Running State Change CS_REPORTING 2011-11-18 15:27:36.633145 [DEBUG] switch_core_state_machine.c:635 (sofia/external/1010@xx.xx.xx.xx:5060) State REPORTING

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SIP INVITE manipulation
by William Alianto 21 Nov '11

21 Nov '11

Hi, I'm looking for a way to manipulate the SIP INVITE message. Currently my SIP server sending 622X to call to provider. But the provider only expect 2X from the invite, which cause the call won't be recognized. Is there any string on diaplan I can add to adjust the invite message, so the provider recognize the call? -- Regards, William

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