November 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Memory problems [SOLVED]
by Babstar 23 Nov '10

23 Nov '10

As alluded to by earlier posters this problem was actually hacking attempts. Although they didn't show up in the logs, I actually had to do a packet capture on the firewall, and it showed a flood of register attempts. I I blocked everything on port 5060 except for my SIP providers, and so far problem solved. Hope this helps someone else. -- Babstar

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Voicemail Email to Multiple addresses
by Aloysius Lloyd 23 Nov '10

23 Nov '10

Hi All, I am trying to send voicemail to multiple email addresses. I use the comma separated list. But it is not sending any emails. But if I use a single email address it is working. <param name="vm-mailto" value="user1@domain.com,user2@domain.com"/> Wiki "Multiple email addresses can be defined using a comma-separated list." Any help appreciated. Thanks Lloyd

2 1

T.38 and Audiocodes or Patton gateway
by Tim St. Pierre 23 Nov '10

23 Nov '10

Hello, I am trying to get a reliable fax setup working. I have set up a nice fax receiver mechanism with mod_spandsp and it works very well when I send a fax using a Linksys SPA2102 I have poor results with Audiocodes MP202B, and can't get anything to work with a Patton M-ATA I have a number of Audiocodes MP202B in service and would rather not replace them if I can avoid it. In each case, the fax fails with "communications error" on the sending machine, and "Call hungup prematurely" on the SpanDSP side. There aren't too many knobs on the gateways. Are there any knobs in mod_spandsp that might help this condition? Anyone else have any experience with these devices? Thanks! -Tim

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Help! FS Unable to Handle Incoming Calls Through SIP Gateway
by Yasuro 23 Nov '10

23 Nov '10

Hi, FreeSWITCH gurus! I need your help! First off, I am new to FS and I am new to Internet telephony as well. Heck, I am new to the concept of NAT, UPnP, etc., so please bear with my ignorance. I subscribe to a VoIP service at home, with which I get one DID. They supply me a VoIP adapter. Their expected usage is for you to plug in analog phones to the analog phone jacks in the VoIP adapter. However, It also has four Ethernet LAN ports and it acts as a router. You can also access it from the LAN side and register with its built-in SIP gateway. What I would like to do is to run FS (Windows version) on one of the Windows PCs, have it register with the SIP gateway, and have it act as an AA or IVR. For testing, I am having it just play music. When I tried the same idea with AsterikWin32, it worked just as I had hoped; it answered incoming calls automatically. However, I somehow cannot make it work with FS. I simulate incoming calls to my DID number with Skype's Sypeout. It fails after a short while with such error messages as "network error." It appears the call was never answered. FS is assigned an extension number 7 at the gateway. When I call extension 7 from a different extension (at the gateway level, not an extension inside FS), FS does answer the call and I hear music. FS fails to answer only incoming calls from outside. I think my FS configuration is fairly standard. I created an external SIP profile for the gateway under conf/sip_profiles/external/ and modified conf/dialplan/public/00_inbound_did.xml so incoming calls to the gateway will be transferred to an extension within FS. FS's messages and logs, plus the result of packet captures indicate that FS /thinks /it has answered the call, and goes on to initiate media communication. I see RTP packets going from FS to the SIP gateway. What's different from AsteriskWin32's case is that there are no RTP packets coming back from the SIP gateway to FS. Turning of the firewall of the PC does not seem to change the result in any way. For your perusal, I have created the following logs of communication between FS/AsteriskWin32 and the SIP gateway: * AsteriskWin32's case o Summary: http://pastebin.freeswitch.org/14457 o Details: http://pastebin.freeswitch.org/14460 * FreeSWITCh's case o Summary: http://pastebin.freeswitch.org/14462 o Details: http://pastebin.freeswitch.org/14463 o Log: http://pastebin.freeswitch.org/14465 The IP address of the SIP gateway is *192.168.11.250*, and that of the PC FS/AsteriskWin32 resides in is *192.168.11.11*. My DID number is masked as ABCDEFGHIJ. I do not know if it gives you any useful information, but those files include the registration phase. /FS's log was taken at a different time/, so it does not entirely match the packet captures. I also have the corresponding Pcap files. Please let me know if you need them. I am not entirely sure, but I think as far as what I'd like to do is concerned, NAT is not going to be an issue, because the FS/AsteriskWin32 PC and the SIP gateway (its LAN side IP address) are on the same subnet (192.168.11/24). At this time, I do not need to access FS from the Internet. Finally, I will give you more details about my setup, which may or may not be relevant to this issue. My home LAN is set up this way: http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetu… Please note that there are /two layers/ of NAT, and that in the inner layer, two NAT devices exist. I know it looks convoluted, but there are logical reasons for this setup. The VoIP service provider only supports the PCMU codec. The music file I prepared for this testing is encoded in PCMU, so codecs will not be an issue. Please do not hesitate to ask if you have any questions. Thanks for your help in advance! Yasuro

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One Trunk not working with FreeTDM Sangoma ISDN library
by Stephen Wilde 22 Nov '10

22 Nov '10

Hi, I have Freeswitch git head revision, configured with FreeTDM module and Sangoma ISDN library with A108 board. The problem I have is that there is one trunk (FreeTDM/9:) where all calls fail with a cause 111. Nothing changes if I restart Freeswitch or I do a wanrouter restart. All other trunks work fine. The freetdm.conf is: http://pastebin.com/XWbUBwhp The freetdm.conf.xml is: http://pastebin.com/XVJpPSL6 The wanpipe9.conf: http://pastebin.freeswitch.org/14589 Tracing the Q931 with wanpipemon what I see is only Q921 RR messages in the two directions, so the SETUP message is not sent to the other end! In the Freeswitch log I see: 2010-11-21 23:17:14.320500 [DEBUG] mod_freetdm.c:1334 Connect outbound channel FreeTDM/9:1/393494255737 2010-11-21 23:17:14.320500 [NOTICE] switch_channel.c:784 New Channel FreeTDM/9:1/393494255737 [cd6c2968-40ee-41c6-bbe5-d7c7c8222c40] 2010-11-21 23:17:14.320500 [DEBUG] mod_freetdm.c:1348 (FreeTDM/9:1/393494255737) State Change CS_NEW -> CS_INIT 2010-11-21 23:17:14.320500 [DEBUG] switch_core_session.c:1064 Send signal FreeTDM/9:1/393494255737 [BREAK] 2010-11-21 23:17:14.320500 [DEBUG] ftmod_sangoma_isdn.c:692 [s9c1][9:1] Changed state from DOWN to DIALING 2010-11-21 23:17:14.320500 [DEBUG] ftmod_sangoma_isdn.c:471 [s9c1][9:1] processing state change to DIALING 2010-11-21 23:17:14.320500 [INFO] ftmod_sangoma_isdn_stack_out.c:142 [s9c1][9:1] Outgoing call: Called No:[393494255737] Calling No:[0872868784] 2010-11-21 23:17:14.320500 [INFO] ftmod_sangoma_isdn_stack_out.c:149 [s9c1][9:1] Sending SETUP (suId:1 suInstId:24 spInstId:0 dchan:9 ces:0) 2010-11-21 23:17:14.321526 [INFO] ftmod_sangoma_isdn_stack_rcv.c:248 [s9c1][9:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:24 spInstId:2157) 2010-11-21 23:17:14.321526 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:511 [s9c1][9:1] Processing RELEASE/RELEASE COMPLETE (suId:1 suInstId:24 spInstId:2157) 2010-11-21 23:17:14.321526 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:552 [s9c1][9:1] cause:111 2010-11-21 23:17:14.321526 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:559 [s9c1][9:1] Changed state from DIALING to TERMINATING 2010-11-21 23:17:14.321526 [DEBUG] ftmod_sangoma_isdn.c:471 [s9c1][9:1] processing state change to TERMINATING 2010-11-21 23:17:14.321526 [DEBUG] ftdm_io.c:4838 [s9c1][9:1] Scheduling safety hangup timer 2010-11-21 23:17:14.321526 [DEBUG] mod_freetdm.c:2049 got clear channel sig [STOP] 2010-11-21 23:17:14.321526 [DEBUG] switch_channel.c:2455 (FreeTDM/9:1/393494255737) Callstate Change DOWN -> HANGUP 2010-11-21 23:17:14.321526 [NOTICE] mod_freetdm.c:2074 Hangup FreeTDM/9:1/393494255737 [CS_INIT] [PROTOCOL_ERROR] 2010-11-21 23:17:14.321526 [DEBUG] switch_ivr_originate.c:3439 Originate Resulted in Error Cause: 111 [PROTOCOL_ERROR] 2010-11-21 23:17:14.321526 [DEBUG] switch_core_state_machine.c:320 (FreeTDM/9:1/393494255737) Running State Change CS_HANGUP 2010-11-21 23:17:14.321526 [DEBUG] switch_channel.c:2471 Send signal FreeTDM/9:1/393494255737 [KILL] 2010-11-21 23:17:14.321526 [DEBUG] switch_core_session.c:1064 Send signal FreeTDM/9:1/393494255737 [BREAK] 2010-11-21 23:17:14.322566 [DEBUG] switch_core_state_machine.c:557 (FreeTDM/9:1/393494255737) State HANGUP 2010-11-21 23:17:14.322566 [DEBUG] mod_freetdm.c:515 [s9c1][9:1] Changed state from TERMINATING to HANGUP 2010-11-21 23:17:14.322566 [DEBUG] ftmod_sangoma_isdn.c:471 [s9c1][9:1] processing state change to HANGUP 2010-11-21 23:17:14.322566 [DEBUG] ftmod_sangoma_isdn.c:564 [s9c1][9:1] Acknowledging remote abort 2010-11-21 23:17:14.322566 [DEBUG] ftmod_sangoma_isdn.c:600 [s9c1][9:1] Changed state from HANGUP to HANGUP_COMPLETE 2010-11-21 23:17:14.322566 [DEBUG] ftmod_sangoma_isdn.c:471 [s9c1][9:1] processing state change to HANGUP_COMPLETE 2010-11-21 23:17:14.322566 [DEBUG] ftmod_sangoma_isdn.c:608 [s9c1][9:1] Changed state from HANGUP_COMPLETE to DOWN 2010-11-21 23:17:14.322566 [DEBUG] ftmod_sangoma_isdn.c:471 [s9c1][9:1] processing state change to DOWN 2010-11-21 23:17:14.322566 [DEBUG] ftmod_sangoma_isdn_support.c:46 [s9c1][9:1] Clearing call data (suId:1 suInstId:24 spInstId:0) 2010-11-21 23:17:14.322566 [DEBUG] ftdm_io.c:2385 channel done 9:1 What can I do do put trunk 9 in working state? Stephen

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Presence api for custom function
by Tóth Róbert 22 Nov '10

22 Nov '10

Hello How can i use the Presence api for my custom application? I would like to set my phone's BLF to set that other phones may or may not be able to pickup the ringing calls. I make a dialplan application for this, and the setting is stored in db. Almost everything is OK, the dialplan application works, BLF's color is changing, but after some seconds it changes back to red. It is obviously because there is not a real phone-ext behind the number of what the phone is subscribing. I thought if i use the presence api, its state does not changes until i send another event for that entity. But it is not true. What can i do? thx

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slow execution of answer in ESL mode
by xuyan yang 22 Nov '10

22 Nov '10

hi, During the development of some Java ESL script with Dvarnes' library, I noticed the execution of "answer" in ESL mode is much slower than in XML dialplan. It take about 1-2 second in the ESL version while in dialplan mode it is a immediate return. This is not a critical problem, but I just fell strange and want to ask whether it is a bug or something else?

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blind transfer context
by François Delawarde 22 Nov '10

22 Nov '10

Hello, When doing blind transfer, the context used by freeswitch is the one of the transferee instead of the transferer like in attended transfer scenarios. I would like it to use the transferer context. The variable force_transfer_context can be used to force this to a specific context. My problem is that force_transfer_context must be set in the transferee channel which makes it quite complicated if for example I bridge to a groups of users that use different contexts (need an answer hook to set that variable), or in chain transfer scenarios (need to re-set the variable to the context of the new bridged user). Wouldn't it be more logical to use the force_transfer_context variable from the channel performing the transfer to have blind transfers behave the same way as attended ones (using the transferer's context)? Thanks, François.

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Installation and Tips for Mac OS X now on Wiki
by Mario G 22 Nov '10

22 Nov '10

My first contribution to FreeSWITCH: http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X. I updated 2 sections of the main installation page as well. I wish I had it for my first OS X installation, hopefully it will prove very useful for people coming to this list for OS X installation help. Critics be gentle, this is my very first wiki page and I spend many, many, many hours/days/weeks on it. Someone commented that an install page is needed for each distribution. I may do OpenSuse in the future if there is interest. Mario G

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OpenBSD
by Brian West 22 Nov '10

22 Nov '10

Anyone wishing to look over this jira: http://jira.freeswitch.org/browse/FS-2863 If not I'm going to close it tomorrow as we don't officially support OpenBSD so someone will need to step up in our community to manage OpenBSD. Thanks, Brian

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