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Re: [Freeswitch-users] Problem with inbound call answered but no sound
by Eliot Gable 27 Oct '09

27 Oct '09

FYI, it generally makes debugging easier if you do this: sofia profile external siptrace on sofia profile internal siptrace on That way you can see the actual signaling and it is usually more clear what is going on. In most cases, you will probably be able to figure it out yourself just looking at the signaling. On Mon, Oct 26, 2009 at 10:29 PM, Lars Zeb <larclap(a)yahoo.com> wrote: > I have tried to update (make current) twice since 15183. All inbound calls > are picked up but the caller hears nothing but a couple of clicks. The most > recent version I’ve tried is 15241. > > > > Any ideas on what may be causing this? > > > > http://pastebin.freeswitch.org/10843 > > > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > Thanks Lars > > _______________________________________________ > 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 > > -- Eliot Gable "We do not inherit the Earth from our ancestors: we borrow it from our children." ~David Brower "I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime." ~David Brower "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero

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mod_nibblebill and memory problem
by Dome Charoenyost 27 Oct '09

27 Oct '09

Dear All, I'm running mod_nibblebill for my prepaid solution. I still have problem with memory. I have 4 GB RAM and runing debian squeeze 64 bit and 200 calls concurrent Last time nibblebill running with 1 min heartbeat. when i check memory by htop FS user memory 2% anf growth to 60-89% in 8-9 hr. and then FS crash. Now i change hearbeat to 0 it's mean nibble update balance when end of call. but everything are same FS start from 2% and growth to 20% in 2 days. When i unload nibblebill FS running fine. My question is when concurrent calls drop to 1-2 calls why FS (I think nibblebill) still use memory ? something wrong in nibblebill ? BG Dome C.

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Setup advice on small LAN
by Lars Zeb 26 Oct '09

26 Oct '09

I am currently running FreeSwitch very successfully, thanks to the help from many on this list. I am new to Linux so it was a challenge. FS runs on a small LAN with about 5 other computers. The connection to the internet is DSL with 3M down and 768kb down via Covad. The ITSP is Flowroute. If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure. Somehow I need to isolate the FS box from the rest of the LAN, or give its traffic precedence. Covad's suggestion was to place the FS box in the DMZ. If I have to, I'll get another DSL line and isolate it that way. Is there a write-up anywhere that might help me with this problem, or lacking that, can anyone offer advice? 581 2009-10-26 07:15:58.307011 [NOTICE] sofia_reg.c:333 Registering flowroute 582 2009-10-26 07:15:59.275622 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected 583 2009-10-26 07:15:59.470887 [DEBUG] sofia_reg.c:1414 Changing expire time to 2243 by request of proxy sip:sip.flowroute.com ... 1087 2009-10-26 07:16:00.623158 [DEBUG] mod_event_socket.c:2302 Socket up listening on 127.0.0.1:8021 1088 2009-10-26 07:16:01.382171 [DEBUG] sofia.c:707 nua_i_outbound: unknown event 8: 101 NAT detected 1090 2009-10-26 07:26:34.918724 [DEBUG] sofia_reg.c:1414 Changing expire time to 1607 by request of proxy sip:sip.flowroute.c 1091 2009-10-26 07:44:56.715662 [DEBUG] sofia_reg.c:1414 Changing expire time to 505 by request of proxy sip:sip.flowroute.co 1092 2009-10-26 07:49:52.278202 [DEBUG] sofia_reg.c:1414 Changing expire time to 210 by request of proxy sip:sip.flowroute.co 1093 2009-10-26 07:52:30.468045 [DEBUG] sofia_reg.c:1414 Changing expire time to 52 by request of proxy sip:sip.flowroute.com 1094 2009-10-26 07:52:49.234006 [DEBUG] sofia_reg.c:1414 Changing expire time to 33 by request of proxy sip:sip.flowroute.com 1095 2009-10-26 07:53:00.605161 [DEBUG] sofia_reg.c:1414 Changing expire time to 22 by request of proxy sip:sip.flowroute.com 1096 2009-10-26 07:53:12.379214 [DEBUG] sofia_reg.c:1414 Changing expire time to 10 by request of proxy sip:sip.flowroute.com 1097 2009-10-26 07:53:19.627029 [DEBUG] sofia_reg.c:1414 Changing expire time to 3 by request of proxy sip:sip.flowroute.com 1098 2009-10-26 07:53:20.106923 [NOTICE] sofia_reg.c:333 Registering flowroute 1099 2009-10-26 07:53:20.454870 [DEBUG] sofia_reg.c:1414 Changing expire time to 2 by request of proxy sip:sip.flowroute.com 1100 2009-10-26 07:53:20.705833 [NOTICE] sofia_reg.c:333 Registering flowroute 1101 2009-10-26 07:53:20.952781 [DEBUG] sofia_reg.c:1414 Changing expire time to 1 by request of proxy sip:sip.flowroute.com 1102 2009-10-26 07:53:21.205740 [NOTICE] sofia_reg.c:333 Registering flowroute 1103 2009-10-26 07:53:21.705656 [NOTICE] sofia_reg.c:333 Registering flowroute 1104 2009-10-26 07:53:22.002609 [DEBUG] sofia_reg.c:1414 Changing expire time to 60 by request of proxy sip:sip.flowroute.com 1105 2009-10-26 09:06:17.554726 [ERR] sofia_reg.c:1425 flowroute Registration Failed with status Operation has no matching challenge [904]. failure #1 Thanks Lars

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Downloaded tar vs latest SVN - 200 OK has more headers
by Ujjval Karihaloo 26 Oct '09

26 Oct '09

Hi, I used the downlaoded TAR ball and my calls worked, however, when upgrading to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the call - using Conferencing app).. Here are teh bad and good 200 OK,,,,I see a lot of additional headers startin gwith X:FS , can I remove them? Bad one ----------------- SIP/2.0 200 OK v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bK1662379c08490b63b0b9be6978d93ebe f: "SIMPLE SIGNAL "<sip:2222220208@x.x.x.20:5080;user=phone>;tag=945d1734da594444aac9d071edc798a7 To: "trunkgroup user" <sip:1111114847@simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a>;tag=gX1Z34952QBpS i: 674babc2dbf2a90da97bd68642865eae-23f978(a)x.x.x.210<mailto:674babc2dbf2a90da97bd68642865eae-23f978@x.x.x.210> CSeq: 3240 INVITE Contact: <sip:1111114847@x.x.x.20:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-FS-Display-Name: +11111114847 X-FS-Display-Number: +11111114847 X-FS-Support: update_display P-Asserted-Identity: "+11111114847" <+11111114847> v=0 o=FreeSWITCH 1256498762 1256498763 IN IP4 x.x.x.20 s=FreeSWITCH c=IN IP4 x.x.x.20 t=0 0 m=audio 29406 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------ Good One ------------------------------- SIP/2.0 200 OK v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bKffebde0efa06a6aa7772bae5ca085627 f: "SIMPLE SIGNAL "<sip:2222220208@x.x.x.20:5080;user=phone>;tag=895fd35d4979e6e5b0cdcc795b8c8df2 To: "trunkgroup user" <sip:1111114847@simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a>;tag=KBXKrBQU99c6D i: 28f0d47f1163e0d478247e6729762d98-23fedb(a)x.x.x.210<mailto:28f0d47f1163e0d478247e6729762d98-23fedb@x.x.x.210> CSeq: 3278 INVITE Contact: <sip:1111114847@x.x.x.20:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 v=0 o=FreeSWITCH 1256512215 1256512216 IN IP4 x.x.x.20 s=FreeSWITCH c=IN IP4 x.x.x.20 t=0 0 m=audio 17376 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20

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problems registering on www.freeswitch.org
by Simon J Mudd 26 Oct '09

26 Oct '09

Hi, I tried to register on freeswitch.org using username sjmudd. It said the username already existed so I requested having my password resent. Nothing recieved. I've also tried to register a new user simon.mudd with the same results, no mail received. I also see the following when trying to register which looks worrying: http://merlin.wl0.org/20091026/e46494be187a53b3c27740aaff360910abd4facb.png I'm pretty sure that I'm not filtering out the emails but can't be sure. Could whoever maintains the web page contact me off list to help me determine where the problem is? Thanks, Simon Mudd

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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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