October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Inbound DTMF Not Recognized By IVR
by Jerry Richards 26 Oct '09

26 Oct '09

I installed FS on a machine with a Sangoma A101D (PRI) card and if I make an inbound call to the FS IVR, it does not recognize DTMF digits from the PSTN phone. If I call IVR from an internal phone, then it does recognize the DTMF digits. I have mostly default configurations for everything. Best Regards, JErry

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Estimating Call Capacity
by Ujjval Karihaloo 26 Oct '09

26 Oct '09

With the following spec for CPU and Memory can someone help me guesstimating how many simultaneous calls and Calls/sec a FS server can handle - Used as a Conferencing Server. cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping : 1 cpu MHz : 2800.386 cache size : 1024 KB physical id : 0 siblings : 2 core id : 0 cpu cores : 1 fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips : 5604.12 processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping : 1 cpu MHz : 2800.386 cache size : 1024 KB physical id : 0 siblings : 2 core id : 0 cpu cores : 1 fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips : 5600.22 # cat /proc/meminfo MemTotal: 1035080 kB MemFree: 517972 kB Buffers: 85812 kB Cached: 271264 kB SwapCached: 0 kB Active: 224292 kB Inactive: 223008 kB HighTotal: 130816 kB HighFree: 29484 kB LowTotal: 904264 kB LowFree: 488488 kB SwapTotal: 2031608 kB SwapFree: 2031520 kB Dirty: 80 kB Writeback: 0 kB AnonPages: 90172 kB Mapped: 39880 kB Slab: 60060 kB PageTables: 3232 kB NFS_Unstable: 0 kB Bounce: 0 kB CommitLimit: 2549148 kB Committed_AS: 345780 kB VmallocTotal: 114680 kB VmallocUsed: 3584 kB VmallocChunk: 110888 kB HugePages_Total: 0 HugePages_Free: 0 HugePages_Rsvd: 0 Hugepagesize: 4096 kB

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Qustion about INFO messages after Connect/Answer
by Helmut Kuper 26 Oct '09

26 Oct '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, after updating FS to trunk a few days ago I found that callee's display is updated serveral times to caller's name after callee picked up. The first two equal INFO messages looks like this: INFO sip:2850@85.16.245.213:1040;line=367hfn9i SIP/2.0 Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bKU4803UHXaKD2r Max-Forwards: 70 From: "Snom1 an PBX" <sip:4918@85.16.246.6>;tag=ccc4B2F9SD2rm To: <sip:2850@85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330 CSeq: 121727215 INFO Contact: <sip:mod_sofia@85.16.246.6:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: message/sipfrag Content-Length: 27 From: To: "unknown" 2850 The third one is this: INFO sip:2850@85.16.245.213:1040;line=367hfn9i SIP/2.0 Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK0H7vc35evZvZj Max-Forwards: 70 From: "Snom1 an PBX" <sip:4918@85.16.246.6>;tag=ccc4B2F9SD2rm To: <sip:2850@85.16.245.213:1040;line=367hfn9i>;tag=zbrirvg9ow Call-ID: 97a0e683-34e6-122d-8db2-00144fe6e330 CSeq: 121727216 INFO Contact: <sip:mod_sofia@85.16.246.6:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: message/sipfrag Content-Length: 32 From: To: "Snom1 an PBX" 4918 What calle sees is this: 1. Phone rings: caller's displayname 2. Callee picks up: switching from dislayname to unknown 3. Switching from unknown to displayname After callee has picked up caller's display is also updated with callee's name (this is a copy of callee's number rather than callee's name) and number. First Question: Can this behaviour be disabled? Or can it be modified by dialplan, so that there is no "unknown"? Unfortunately when you do an outgoing call via openzap and callee picks up the phone caller's display is updated with two equal INFO message like this: INFO sip:2850@85.16.245.213:1040;line=367hfn9i SIP/2.0 Via: SIP/2.0/UDP 85.16.246.6;rport;branch=z9hG4bK7jFa9Ztg39UUr Max-Forwards: 70 From: <sip:890327@85.16.246.6;user=phone>;tag=9QZamXtmUyXvm To: "Helmut Kuper" <sip:2850@85.16.246.6>;tag=5lf2e0q1on Call-ID: 3c2a2539a966-0m3i9s85my46 CSeq: 121727707 INFO Contact: <sip:890327@85.16.246.6:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15137M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: message/sipfrag Content-Length: 36 From: To: "1/a/890327" 1/a/890327 Second Question: Can I change the name and number via dialplan, so that the correkt name and number is viewed to caller? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK2Fmk4tZeNddg3dwRApPUAKCQLBEi/l6mwf1c55r5mlnEqvLLowCgoP9t VSQFKUCWpsxVm3evyfWQT/Y= =FGsp -----END PGP SIGNATURE-----

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Mod_socket: custom caller sip domain in originate command
by Artem Shiyanov 26 Oct '09

26 Oct '09

Hi there! Please, suggest how to specify custom caller sip domain (logical) in originate command. I've been trying several alternatives but no one worked: 1) specify full sip address in origination_caller_id_number=1000(a)uat.pbx.mblagov.starpoundtech.net - FS adds its IP address so the result From address would be 1000@uat.pbx.mblagov.starpoundtech.net@ 172.26.200.250 2) specify sip_from_uri varuable like follows: bgapi originate {origination_caller_id_number=1000, sip_from_uri=1000@uat.pbx.mblagov.starpoundtech.net,originate_timeout=10}[origination_uuid=bf8e9778-d410-4504-988d-2d405303183c]sofia/internal/1002%172.26.200.250 &park() FS doesn't even dial the called endpoint. Thanks, Artem

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Reroute a Call Based on the Disconnect Cause
by Delian Tashev 26 Oct '09

26 Oct '09

Hello VoIP Geeks! I am looking for a way to reroute calls on specific disconnect reasons. My application is in Python. I will provide my solves so far, which for some reason I cannot use. Why I need rerouting? Well, I am using the LCR module which provides multiple routes per destination, like this: routes= [lcr_carrier=fs-rocks.com,lcr_rate=0.15000]sofia/default/1174(a)194.135.34.200 :5062| [lcr_carrier=fs-rocks.com,lcr_rate=0.16000]sofia/default/1174(a)194.135.34.200 :5064 Solve 1: Python solve Split the routes string to multiple routes. Bridge the call. Check if the call failed and bridge again. ivr_reroute_on=(\ "CALL_REJECTED",\ "CRASH",\ "INVALID_NUMBER_FORMAT",\ "NORMAL_TEMPORARY_FAILURE",\ "NO_ROUTE_DESTINATION",\ "NO_USER_RESPONSE",\ "NUMBER_CHANGED",\ "RECOVERY_ON_TIMER_EXPIRE",\ "SERVICE_NOT_IMPLEMENTED",\ ) ... session.execute("bridge", route) hangup_cause = str(session.getVariable("originate_disposition")) if hangup_cause in ivr_reroute_on: #bridge again ... else: ... #do not bridge again This is tested and works. However my manager does not likes it. I am not commenting on the side effects of this code (i.e. if you have some timers/schedules/scheduled events etc). Solve 2: Freeswitch Source Code solve Modify the source code of "switch_ivr_originate.c": if (to && !oglobals.continue_on_timeout) { goto outer_for; } after line 2523: +if (switch_channel_cause2str(*cause) == SWITCH_CAUSE_CALL_REJECTED ) { + goto outer_for; +} (In the best case I will add a new channel variable to Freeswitch to control all the disconnect reasons from here) However my manager again does not likes this. He says that I am not good enough in C/C++ to use this solution. Solve 3: continue_on_fail I have tried using the channel variable "continue_on_fail" which I believe works only in the dialplan with no luck. I was unable to get it running both in the dialplan and in Python. By the way I saw that "continue_on_fail" is indeed implemented elsewhere, while "continue_on_timeout" is exactly in "switch_ivr_originate.c" as expected. Solve 4: So can you please people recommend me any other solution? Any function or variable to do the trick? Thank you for your time and help! Delian Tashev

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INFO-Messages: Send_display in pre_answer state doesnt work anymore
by Helmut Kuper 26 Oct '09

26 Oct '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, when I use pre_answer app and right after that the send_display app, there is no INFO message anymore. My dialplan condition: <condition field="${sip_hangup_phrase}" expression="^Do Not Disturb" break="on-true"> <action application="pre_answer"/> <action application="send_display" data="Bitte nicht stoeren"/> <action application="sleep" data="500"/> <action application="playback" data="ansagen/22.wav"/> </condition> changing "pre_answer" to "answer" works, but I don't want to answer the call for error messages. I use latest SVN trunk. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFK4Z0O4tZeNddg3dwRAkB/AKCLN9luR0KGunHVGso2t3wJIo4dqQCdGdjp CY+yWGyVlyDpDs//WFtCqiA= =eG4W -----END PGP SIGNATURE-----

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Hostname
by freeswitch noob 26 Oct '09

26 Oct '09

I have mod xml_curl installed and I am getting the following passed to my script. [hostname] => myhost.local [section] => dialplan I also have multiple versions of FS running on the same box. Is there a way to have each FS instance on my box have a unique hostname ? Thanks in advance.

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sched_api doesn't get launched
by Henry Huang 26 Oct '09

26 Oct '09

Hi: I am using mod_java. And in my script I was able to achieve using: execute("sched_hangup", "+300 alloted_timeout"); However, when I try to run sched_api in the same way, system log returns that it's an invalid application. I have also tried to trigger it with many conditional channel variable api calls , but non of them seemed to execute the api command (because I turned on the highest level of debugging and see no where the sched_api is being called. The closest thing I got was by using "api_after_bridge" like the following, but it only launches when the bridge is teared down(which is not what I want). I originally thought after bridge means right after the 2 party is connected. All I want is to be able to play some message to leg A at certain time. setVariable("api_after_bridge", "sched_api +10 none uuid_displace ${uuid} start /path/to/some.wav 20 mux"); I have been struggling with different combination for a week now.. Please shed some light if you know something. Thanks, -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant

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Database for Audio Data
by Pajongjit Buntaokit 25 Oct '09

25 Oct '09

Hi, Does anyone know whether FreeSWITCH has a function to automatically record every call as an audio file in a server or forward them to be stored in a database with additional parameters such as caller ID, date, starting time and ending time? So that these recorded audio data can be queried and retrieved with the caller ID, date and time. Any suggestion or guidance, please advise. Thank you very much! _________________________________________________________________ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/

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sip chat question
by Juan Backson 25 Oct '09

25 Oct '09

Hi, I am not able to send chat message from the console and the error I am getting is a strange "Invalid Profile". Does anyone know what is wrong? freeswitch(a)server.com> chat sip|454047001854840(a)192.168.1.87| 454047001854840(a)192.168.1.87|testing API CALL [chat(sip|454047001854840@192.168.1.87|454047001854840@192.168.1.87|testing)] output: Error! Message Not Sent freeswitch(a)server.com> 2009-10-25 10:51:01.793313 [ERR] sofia_presence.c:93 Chat proto [dp] from [454047001854840(a)192.168.1.87] to [454047001854840(a)192.168.1.87] testing Invalid Profile 192.168.1.87 freeswitch(a)server.com> freeswitch(a)server.com> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia@192.168.1.87:5080 RUNNING (0) internal profile sip:mod_sofia@192.168.1.87:5060 RUNNING (0) ================================================================================================= 2 profiles 0 aliases

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