June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Spandsp events not being fired
by Joshua Nankin 21 Jun '12

21 Jun '12

I'm sending a fax with freeswitch, but the events listed at http://wiki.freeswitch.org/wiki/Mod_spandsp#Events do not seem to be fired. Do I need to enable them somehow?

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Spandsp - Cannot send non-existant fax file (permissions?)
by Carlo Dimaggio 21 Jun '12

21 Jun '12

Hi all, I'm in trouble with mod_spandsp and freeswitch executed by user "freeswitch". Even if the file is present and the permissions are ok (readable by everyone), the FS console gets: 2012-06-19 16:06:30.098112 [ERR] mod_spandsp_fax.c:1289 Cannot send non-existant fax file [/usr/local/freeswitch/storage/fax/test_fax.tif] FS is able to send if I execute the main process from user "root". Does spandsp require other special permissions? Regards,

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follow-up: mod_python build error
by Eugene Shcherbatyuk 21 Jun '12

21 Jun '12

Sorry for missing these bits (according to http://wiki.freeswitch.org/wiki/Report_Issue_Checklist) root@fs:/usr/local/src/freeswitch# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 30 model name : Intel(R) Core(TM) i5 CPU 760 @ 2.80GHz stepping : 5 cpu MHz : 2904.327 cache size : 6144 KB 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 mce cx8 apic mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 constant_tsc up pni monitor ssse3 bogomips : 5808.65 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: root@fs:/usr/local/src/freeswitch# uname -a 72-Ubuntu SMP Fri Jul 29 21:08:37 UTC 2011 i686 GNU/Linux Yes, situation was reproducible: a) when updating b) when trying to make fresh build. c) when trying to build additional module No, no modification to build scripts, except for uncommenting mod_python in modules.conf > Today I could not update mod_python after git pull. Well, I am not quite certain what to do next except for reporting to the list... > > Best regards, > Gene > > OS: ubuntu 10.04 LTS server > > FreeSWITCH: Version 1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T105634Z > > Message: > > root@fs:/usr/local/src/freeswitch# make mod_python > > making all mod_python > make[4]: *** [mod_python_wrap.cpp] Error 1 > make[3]: *** [all] Error 1 > make[2]: *** [mod_python-all] Error 1 > make[1]: *** [mod_python] Error 2 > make: *** [mod_python] Error 2 > root@fs:/usr/local/src/freeswitch# ========================================================= Данное сообщение и любые вложения («Сообщение») являются конфиденциальными, предназначенными исключительно для адресатов, и могут содержать сведения, охраняемые в соответствии с законодательством. Любое несанкционированное использование или распространение Сообщения запрещено. Электронная почта не гарантирует подлинности Сообщений. Ни ЗАО "АКБ "БЕЛРОСБАНК", ни любые его подразделения или дочерние организации не несут ответственности за Сообщение в случае его подмены, изменения либо фальсификации. ========================================================= This message and any attachments (the “message”) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC “BELROSBANK” nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified. =========================================================

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Speccing server for enterprise FS
by Alex 21 Jun '12

21 Jun '12

Looking to setup a new Freeswitch server that will be doing quite a lot of traffic. Trying to get a rough idea of where to spend the money (RAM? CPU? Network access?). There will be perhaps 10,000 simultaneous SIP registrations Maybe 500 simultaneous calls, some of (say 50%) which will require a-law to GSM transcoding. I know it's a tricky one to answer, but based on experiences here, which bits tend to glow first?

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Re: [Freeswitch-users] gateway registration using TLS
by Mi Ke 21 Jun '12

21 Jun '12

Hi Sean, It seems that setting realm port to 5061 does not set registration mode to TLS - this is how remote server sees our reg attempt: REGISTER sip:server_ip:5061;transport=tcp SIP/2.0 Via: SIP/2.0/TCP our_ip;branch=z9hG4bKXK27gKtt61X9a Max-Forwards: 70 From: <sip:xxxx@server_ip:5061>;tag=Q4ytjXN7t319Q To: <sip:xxxx@server_ip:5061> Call-ID: 29e638e8-ba6d-11e1-aa8e-e35fbde939c6 CSeq: 29804746 REGISTER Contact: <sip:gw+xxxx@our_ip:5060;transport=tcp;gw=xxxx> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 Instead of switching to TLS mode FS just sends plain unencrypted reg request to remote port 5061. After that remote server closes connection. Have you practically succeded in placing FS into TLS registration mode? Cheers / Mike ----- Original Message ----- From: Sean Devoy Sent: 06/20/12 10:27 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] gateway registration using TLS NO, sorry. I just saw “remote_ip_here” in the error message and thought may you actually had used that. Sorry. From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Mi Ke *Sent:* Wednesday, June 20, 2012 12:20 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] gateway registration using TLS Dear Sean, It's not that bad - remote_ip, xxx and yyy are only placeholders in my example - actual config has real params ;) Did you mean that FS will automatically use TLS for registration when I append :5061 to realm value ? Thanks / MiKe ----- Original Message ----- From: Sean Devoy Sent: 06/20/12 07:03 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] gateway registration using TLS Mi Ke, You should put your actual remote provider’s domain name or tip address in where it says “remote_ip_here” and I assume you have entered you actual username and password where it says “yyy” and “zzz”. It should look SIMILAR to this, but with your provider: <param name="realm" value="sip.voipinnovations.com:5061"/> Hope that helps, Sean From: freeswitch-users-bounces(a)lists.freeswitch.org %5Bmailto:freeswitch-users-bounces@lists.freeswitch.org%5D *On Behalf Of *Mi Ke *Sent:* Wednesday, June 20, 2012 11:46 AM *To:* Freeswitch Users *Subject:* [Freeswitch-users] gateway registration using TLS Hi All, Is it possible to explicitly tell sofia to register as a client using TLS? Here's my TLS-related part of sofia config: <profiles> <profile name="external"> <domains> <domain name="all" alias="true" parse="false"/> </domains> <gateways> <gateway name="xxx"> <param name="register" value="true"/> <param name="username" value="yyy"/> <param name="realm" value="remote_ip_here:5061"/> <param name="password" value="zzz"/> <param name="register-transport" value="tcp"/> <param name="expire-seconds" value="3600"/> <param name="retry-seconds" value="60" /> </gateway> </gateways> <settings> ... <param name="context" value="external"/> <param name="sip-port" value="5060"/> <param name="tls" value="true"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="contact-params" value="tport=tcp"/> <param name="tls-version" value="sslv23"/> <param name="tls-cert-dir" value="/usr/local/freeswitch/conf/ssl"/> <param name="tls-sip-port" value="5061"/> <param name="dialplan" value="XML"/> ... </settings> </profile> </profiles> But that does not work... tport_connected( 0x7f9b44014110): events CONNECTED tport_send_event(0x7f9b44014110) - ready to send to (tcp/remote_ip_here:5061 http://212.58.166.54:5061 ) tport_vsend(0x7f9b44014110): 600 bytes of 600 to tcp/remote_ip_here:5061 http://212.58.166.54:5061 tport_vsend returned 600 tport_set_events(0x7f9b44014110): events IN tport(0x7f9b44014110): reset timer tport_wakeup(0x7f9b44014110): events IN HUP ERR tport_release(0x7f9b44014110): 0x7f9b4401a460 by 0x7f9b4402ca70 with (nil) nta: REGISTER (29733430): Connection reset by peer (104) with tcp/[remote_ip_here]:5061 nua(0x7f9b440196a0): event r_register 503 Service Unavailable Any ideas what I'm doing wrong? Thanks / MiKe

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Re: [Freeswitch-users] Rtp traffic
by Juan Pablo L. 21 Jun '12

21 Jun '12

Just a little clarification, i had a typo and I meant csoft Instead of msoft. -----Original Message----- From: Juan Pablo L. Sent: 21 Jun 2012 12:42:10 GMT To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Rtp traffic Oh thank you very much, i thought i had to configure it somewhere, i will try. Thanks a lot. -----Original Message----- From: Paul Cupis Sent: 21 Jun 2012 08:18:25 GMT To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Rtp traffic On Wed, Jun 20, 2012 at 03:28:45PM -0600, Juan Pablo L. wrote: > Hi, i m trying to connect (establish sip trunks) to huawei MSC (msoft), > so i m requesting the information to configure a gateway on the FS side > for outbound and inbound, but they are requesting that the RTP traffic > be sent to one IP (to the UMG) and the SIP traffic to a differentIP. > I have been reading documentation but i have not found how to > accomplish this. can someone please give me a hint ? is this possible ? > anyone with experience integrating to a huawei msc? This shouldn't be a problem - you just setup the SIP side of it and the RTP should be automatic (the SIP signalling will specify the RTP address/port). Obviously if you are firewalling RTP you will need to allow the traffic through your firewall, but you don't need any special configuration on the FreeSWITCH side to do this. Regards, _________________________________________________________________________ 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 Join Us At ClueCon - Aug 7-9, 2012 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 _________________________________________________________________________ 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 Join Us At ClueCon - Aug 7-9, 2012 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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Re: [Freeswitch-users] Does mod_fifo not take in effective_caller_id_name?
by Eric Sander 21 Jun '12

21 Jun '12

Any plans to export effective_caller_id_name into mod_fifo or is this task considered a bounty? eric_s On Saturday, June 16, 2012 at 1:37 AM, Eric Sander <esander83(a)hushmail.com> wrote: > >Any body know if effective_caller_id_name can be passed on to >modfifo. Read the wiki on it and there's no info. This would be a >helpful feature to include as other parts of FS use this to >display info. > >thanks >eric_s > >On Thursday, June 14, 2012 at 6:35 AM, Andrew Cassidy ><andrew(a)cassidywebservices.co.uk> wrote: >> >>mod_callcenter also ignores those variables unless you set one >>called >>cc_export_vars. Perhaps there's a similar mechanism for >>mod_fifo? >> >>On 14 June 2012 05:31, wrote: >> Hi All, >>I have this basic dialplan: >>I can see the call get logged via the first extension but >it's >>not >>passing the info to modfifo. I can remove the 2nd extension and >>put a >>basic: >>and the caller id name info is displayed on the phones but not >when >>routing through mod_fifo. I have tried both set/export and it >>still >>doesn't seem to work. >>Is this correct? >>Thanks,eric_s >> >>__________________________________________________________________ >_ >>______ >> 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 >> Join Us At ClueCon - Aug 7-9, 2012 >> 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 >>-- >>Andrew Cassidy BSc (Hons) MBCS SSCAManaging Director >> >> T 03300 100 960 F 03300 100 961 E >>andrew(a)cassidywebservices.co.uk W >>www.cassidywebservices.co.uk > > >___________________________________________________________________ >______ >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 > >Join Us At ClueCon - Aug 7-9, 2012 > >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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Re: [Freeswitch-users] Rtp traffic
by Juan Pablo L. 21 Jun '12

21 Jun '12

Oh thank you very much, i thought i had to configure it somewhere, i will try. Thanks a lot. -----Original Message----- From: Paul Cupis Sent: 21 Jun 2012 08:18:25 GMT To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Rtp traffic On Wed, Jun 20, 2012 at 03:28:45PM -0600, Juan Pablo L. wrote: > Hi, i m trying to connect (establish sip trunks) to huawei MSC (msoft), > so i m requesting the information to configure a gateway on the FS side > for outbound and inbound, but they are requesting that the RTP traffic > be sent to one IP (to the UMG) and the SIP traffic to a differentIP. > I have been reading documentation but i have not found how to > accomplish this. can someone please give me a hint ? is this possible ? > anyone with experience integrating to a huawei msc? This shouldn't be a problem - you just setup the SIP side of it and the RTP should be automatic (the SIP signalling will specify the RTP address/port). Obviously if you are firewalling RTP you will need to allow the traffic through your firewall, but you don't need any special configuration on the FreeSWITCH side to do this. Regards, _________________________________________________________________________ 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 Join Us At ClueCon - Aug 7-9, 2012 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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Sofia recover and wrong Sip fields when sending re-invite
by Grygoriy Dobrovolskyy 21 Jun '12

21 Jun '12

Hello list I was able to setup HA cluster with two FS boxes, and if my primary crashed sofia recover is triggered on the second box. However i have noticed that second FS box sending the Re-Invite incorrectly. My FS boxes setup is extremely simple, when i receive a call i forward it to the Opensips proxy, so i use bridge like this: (public dialplan) <condition field="${sip_to_user}" expression="(123456789).*$" break="never"> <action inline="true" application="eval" data="URI=${sip_to_user}"/> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="{absolute_codec_string=PCMA}sofia/internal/987654333(a)17.2.128.75"/> </condition> So what FS does, its sends the INVITE to the opensips like this: (stripped) INVITE sip:987654333@17.2.128.75 SIP/2.0 Via: SIP/2.0/UDP 17.2.128.147;rport;branch=z9hG4bK5N320Bvar00ZN Max-Forwards: 68 From: "Grigoriy Dobrovolskyy" <sip:33811740@17.2.128.147>;tag=3jNgatZH4D92Q To: <sip:987654333@17.2.128.75> ------stripped----------- The call is answerred, and all is going fine, i crash this box and trigger box2 to reinvite the call, FS2 gets the same ip and send Re-Invite like this: INVITE sip:17.2.128.75:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 17.2.128.75;branch=z9hG4bK62a7.6041d237.0;rport Max-Forwards: 68 From: <sip:987654333@17.2.128.75>;tag=B2B.511.202 To: Grigoriy Dobrovolskyy <sip:33811740@17.2.128.147>;tag=3jNgatZH4D92Q Notice the FROM and To field, we are sending the invite to a proxy, from this proxy, this has no sence, it should be: From: Grigoriy Dobrovolskyy <sip:33811740@17.2.128.147>;tag=3jNgatZH4D92Q To: <sip:987654333@17.2.128.75>;tag=B2B.511.202 Is there any way i can fix this with some kind of param ? Greg

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mod_python build error
by Eugene Shcherbatyuk 21 Jun '12

21 Jun '12

Today I could not update mod_python after git pull. Well, I am not quite certain what to do next except for reporting to the list... Best regards, Gene OS: ubuntu 10.04 LTS server FreeSWITCH: Version 1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T105634Z Message: root@fs:/usr/local/src/freeswitch# make mod_python making all mod_python make[4]: *** [mod_python_wrap.cpp] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_python-all] Error 1 make[1]: *** [mod_python] Error 2 make: *** [mod_python] Error 2 root@fs:/usr/local/src/freeswitch# ========================================================= Данное сообщение и любые вложения («Сообщение») являются конфиденциальными, предназначенными исключительно для адресатов, и могут содержать сведения, охраняемые в соответствии с законодательством. Любое несанкционированное использование или распространение Сообщения запрещено. Электронная почта не гарантирует подлинности Сообщений. Ни ЗАО "АКБ "БЕЛРОСБАНК", ни любые его подразделения или дочерние организации не несут ответственности за Сообщение в случае его подмены, изменения либо фальсификации. ========================================================= This message and any attachments (the “message”) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC “BELROSBANK” nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified. =========================================================

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