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Internal profile startup fails if ssl/tls enabled
by Farhan Husain 11 Nov '10

11 Nov '10

Hello, I enabled ssl in my FS installation. Here are few lines from my vars.xml:  <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1"/>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="*internal_ssl_enable=true*"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/> Then I restarted my server and found the following errors: 2010-11-10 20:22:37.555517 [ERR] sofia.c:1446 Error Creating SIP UA for profile: internal-ipv6 2010-11-10 20:22:37.576362 [ERR] sofia.c:1446 Error Creating SIP UA for profile: internal The "sofia status" command does not show the internal and internal-ipv6 profiles. This wiki page: http://wiki.freeswitch.org/wiki/SIP_TLS#If_your_SIP_profile_fails_to_start_… talks about exactly this problem and it suggest that libssl-dev is not installed. But when I checked my Ubuntu 10.04 I found it to be installed. I also installed "libcurl4-openssl-dev" and then did "make; make clean; make install" and reran the server. However, the problem persists. Can anyone tell me what else may be wrong with my server? Thanks, Farhan

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Wiki Questions
by Peder 11 Nov '10

11 Nov '10

Are there any guidelines as to what is acceptable as far as editing other people's work in the wiki? For example, on this page: http://wiki.freeswitch.org/wiki/Getting_Started_Guide If you go down to Command Line Parameters, it shows about 10 options. Right below it, it says "Recent options" and lists the 10 above, plus another 15 or so. So there is redundant information and it is confusing. It should all be put in one item. I don't mind editing when I see stuff like that, I just don't want to step on other people's toes. Also, under Starting in Unix-based Systems, it shows a block of code with a little computer symbol next to it. Then there is a block of code below it with no symbol. Then a block of code below it with a symbol again. Then way down at the bottom, there is a block of code with a pencil next to it. And a block of code with a light bulb next to it.. I'm sure the response will be "it's all yours, go ahead and fix it". I am just curious if there are guidelines, or if anybody looks over changes to make sure they look reasonable. Peder

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Extracting headers from REFER in dialplan?
by Marc Lewis 11 Nov '10

11 Nov '10

Is there any way to extract additional headers from the dialplan when FS receives a REFER message? The application I'm working on takes an incoming call, bridges it to another SIP server. The SIP server does some things with the call, then sends a REFER message back to FS with an extra header that I would like to use in the dialplan. The header shows in a trace, but not in any of the variables that I have access to. For example, the REFER message looks something like this (numbers and addresses changed to protect the innocent): REFER sip:gw+testserver@192.168.1.95:5060;tport=tcp;transport=tcp;gw=testserver SIP/2.0 FROM: <sip:4255551212@192.168.1.155>;epid=D4A38B23EF;tag=81a8713cf TO: <sip:testserver@192.168.1.155;transport=tcp>;tag=UNBQ363BgNvHa CSEQ: 1 REFER CALL-ID: b1aaff2e-6869-122e-53b9-001b216b53af MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 192.168.1.155:5060;branch=z9hG4bK606d7bc CONTACT: <sip:testing01:5060;transport=Tcp;maddr=192.168.1.155;ms-opaque=eaed293d6f65e314>;automata CONTENT-LENGTH: 0 REFER-TO: <sip:14255551213@192.168.1.95> REFERRED-BY: <sip:4255551212@192.168.1.155> X-Information: my dog has fleas I would then like to have something like this in the 4255551213 extension: <action application="eval" data="${chat(jingle|testing@fs.jabber.server|agent@another.jabber.server|${caller_id_number} to Queue - ${variable_sip_h_X-Information})}"/> The problem is the "X-Information" header is not exported by sofia as far as I can tell. Is there any way to get this variable via the XML dialplan? Thanks in advance. - Marc

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Using Lua Dialplan hunting
by Tim St. Pierre 11 Nov '10

11 Nov '10

Hello, I am using lua extensively for dynamic dialplan generatian. In some cases, a dialplan is run with execute_extension instead of transfer. How do I get the Caller-Hunt-Destination from within the lua script? It doesn't seem to be a session variable, and I can't pass arguments to the lua script when it is used as a dialplan. session:getVariable(caller-destination) returns the caller destination, which is usually different than the hunt-destination when execute_extension is used. Any suggestions? Can this be added as a session variable somehow?

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Help! FS Unable to Handle Incoming Calls Through SIP Gateway
by Yasuro 11 Nov '10

11 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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Free SIP client with TLS support
by Farhan Husain 11 Nov '10

11 Nov '10

Hello, Is there any free SIP client other than SIP Communicator which supports SIPS over TLS? I tried to make SIP Communicator work but failed. I know the old Windows Messenger has support for SIP over TLS but Microsoft does not provide it in its download center any more. Thanks, Farhan

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FTDM Sangoma problem with latest git release
by Stephen Wilde 11 Nov '10

11 Nov '10

2 1

FreeTDM - Screening Indicator Field
by Stephen Wilde 11 Nov '10

11 Nov '10

Hi, I have a server with Freeswitch (latest GIT revision) + FreeTDM with some Sangoma A108 boards. I have the necessity to set, for outgoing calls, the ISDN information element "Screening Indicator" of calling party number. I have see that some information element can be set in the "freetdm.conf.xml" file (i.e. calling and called typer of number or numbering plan) but I have not found any reference to the screening indicator. If there is no way to specify this information element for each trunk, it's possible to change its default value (that is 01=user provided, verified and passed)? Thanks, Stephen

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After Call Work
by Bill Kunyiha 11 Nov '10

11 Nov '10

Is it possible to call a lua script after a call to for example write billing records? What’s the best way of doing this?

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Help! FS Unable to Handle Incoming Calls Through SIP Gateway
by Yasuro 11 Nov '10

11 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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