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Absolute conditions are not changing the proceed variable... (mod_dialplan_xml.c)
by Rafael Vargas 01 Feb '12

01 Feb '12

Hi, I've noticed that when I have an extension containing only an absolute condition, the proceed variable in mod_dialplan_xml.c:parse_exten(...) is not changed and the next extension is also executed, thus ignoring the first extension's "continue" attribute, which was set to "false". I'm referring to this code: (mod_dialplan.c:221 in today's GIT) ... } else if (time_match == -1) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session), SWITCH_LOG_DEBUG, "Dialplan: %s Absolute Condition [%s] match=%s\n", switch_channel_get_name(channel), exten_name, all ? "all" : "any"); pass++; if (!all && !xor) break; } else if (time_match == 1) { ... Shouldn't this set the proceed variable to 1, just like the regex condition a few lines above (line 208 in today's GIT)? I just made a quick test and it worked as (I) expected. What I'm proposing is to change the code above to the following: ... } else if (time_match == -1) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG_CLEAN(session), SWITCH_LOG_DEBUG, "Dialplan: %s Absolute Condition [%s] match=%s\n", switch_channel_get_name(channel), exten_name, all ? "all" : "any"); proceed = 1; pass++; if (!all && !xor) break; } else if (time_match == 1) { ... Sorry if this was already discussed, I've searched this list and the users list to no avail... Best regards, Rafael Scheffer Vargas desenvolvimento KHOMP - 15 Anos R. Joe Collaço, 253 www.khomp.com.br Florianópolis | SC +55 48 3722-2916 | 9612-0276 | USA: +1 213-943-1535 vargas(a)khomp.com.br

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get_trillium_val() loop termination
by Juraj Fabo 01 Feb '12

01 Feb '12

Hi I'm using freetdm from git downloaded on 30.1.2012. On windows I had no problems, but on sles 11 sp1 64bit call setup message was missing caller and called information elements. I've tracked down the problem to file src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c, into function get_trillium_val() Particularly, when get_trillium_val(ton_codes, 0, IN_TON_UNK) is called, 0xe6 is returned (which is from the nsf_spec_codes[] table) I think the loop should have some strict terminate condition but not sure about the write intention. My workaround to be able to continue development is following diff -urN src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support_mod.c --- src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support.c 2012-02-01 14:02:30.000000000 +0000 +++ src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn_support_mod.c 2012-02-01 14:02:08.000000000 +0000 @@ -106,10 +106,11 @@ static uint8_t get_trillium_val(ftdm2trillium_t *vals, uint8_t ftdm_val, uint8_t default_val) { ftdm2trillium_t *val = vals; - while(val++) { + while(val) { if (val->ftdm_val == ftdm_val) { return val->trillium_val; } + val++; } return default_val; } @@ -117,10 +118,11 @@ static uint8_t get_ftdm_val(ftdm2trillium_t *vals, uint8_t trillium_val, uint8_t default_val) { ftdm2trillium_t *val = vals; - while(val++) { + while(val) { if (val->trillium_val == trillium_val) { return val->ftdm_val; } + val++; } return default_val; } Is there some other way how to go around? Thank you Juraj

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(no subject)
by Андрей Аппалонов 01 Feb '12

01 Feb '12

Hello, tell, why FS can not hear that the fax has lifted a receiver (I use ventafax)?

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switch_stun_random_string returns identical strings on two calls randomly
by Seven Du 01 Feb '12

01 Feb '12

Hi, I tried to replace the following function which I copied somewhere in msrp.c with switch_stun_random_string (http://docs.freeswitch.org/group__stun1.html#geb124530a232d65edc508afd1b9dd…) but I found sometimes switch_stun_random_string (http://docs.freeswitch.org/group__stun1.html#geb124530a232d65edc508afd1b9dd…) returns identical strings on two sequential calls, with the following function I never found, so I still use it. By looking the code I understand srand seed is time based so it's possible to return duplicate random string, but I found switch_stun_random_string (http://docs.freeswitch.org/group__stun1.html#geb124530a232d65edc508afd1b9dd…) used in some places other than stun, just curious would it cause problems? void random_string(char *buf, switch_size_t size) { long val[4]; int x; for (x = 0; x < 4; x++) val[x] = random(); snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]); *(buf+size) = '\0'; } Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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Update on Brian's Condition
by Michael Collins 31 Jan '12

31 Jan '12

Hey all, Brian says that the latest is as follows: The lump in his chest is actually a lymph node. They are running further tests to see if they can track down what's going with it. The mass in his left lung appears just to be an infection, but again they are running more tests to make sure. Please everyone keep Brian in your thoughts and prayers. Thanks! -Michael

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msrp support
by Seven Du 31 Jan '12

31 Jan '12

Hi, We are working on msrp support in FS. With the following patch to lib sofia, the msrp SDP can deliver to mod_sofia http://pastebin.freeswitch.org/17944 questions: 1) should I submit this to our jira or to the the sofia team? Any suggestions on the patch? 2) As FS is a B2BUA, so I think it would be endpoint-to-endpoint bridge, then we need start TCP/MSRP thread to handle media a. It should more like video handling or more like fax ? That means to hack bridge code to also handle msrp, or should we implement api like tx_fax/rx_fax vs tx_msrp/rxmsrp? b. for text over msrp, is it possible to routing the chatplan like mod_sms c. the first stage would be just endpoint to endpoint, shoud we implement relay ? or even switch mode for conference? I'm very new to msrp. 3) we found a msrp lib from http://confiance.sourceforge.net/ . It's GPL, I know it's not license compatible, but as I learned from previous talk about video, the code can put elsewhere and we just implement the logic to hook them on. Or other lib we could use? Any suggestion is appreciated. Seven. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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UPDATE: Donating To Assist Brian West
by Michael Collins 30 Jan '12

30 Jan '12

Good news, We've contacted PayPal and made sure that they are aware of the fact that Brian may be receive a number of donations. Sometimes this can be construed as "suspicious activity" so it's good to be prepared. PayPal has requested that we use payment type "Personal" and that the money is being sent as a "gift." Here's a quick screenshot: Let's all give Brian some PayPal love! Thanks, Michael P.S. - If you have any questions please email me off list.

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Re: [Freeswitch-dev] ftdm_call_place and caller_data.call_id
by Juraj Fabo 30 Jan '12

30 Jan '12

Hi, Thank you. > > > Hi, >> >> >> You are better off using mod_freetdm.c as the reference, the sample/test > applications may be outdated. > > Unfortunatelly in mod_freetdm.c there seems to be no need for the call_id from the ftdm_call_place_ex(..). > ftdm_channel_t is an opaque type to API users. I don't remember what your > exact question was. You should not use fchan.caller_data, you always get a > copy of caller_data when you need it, the fact that is internally attached > to the fchan pointer is not something you should rely on (and that's why is > in a private header). > > I though that the freetdm generates a new call ID (in context of freetdm) when new call is initated via ftdm_call_place_ex(..) and I was looking for the way how this call_id is announced to the caller application. I've recently tested head from freeswitch git, but the caller_data.call_id is still zero after returning from ftdm_call_place_ex(). My problem is that the comment aroung _ftdm_call_place() in the freetdm api says: * \note When this function returns FTDM_SUCCESS, the member .fchan from caller_data will be set to the channel used to place the call * and .call_id to the generated call id for that call * but my applciation still receives zero. mod_freetdm seems to not care about this call_id at all. So my workaround is stil following: ftdmret = ftdm_call_place_ex(&caller_data, &hunting, usrmsg); if (ftdmret == FTDM_SUCCESS) { callerData = ftdm_channel_get_caller_data(caller_data.fchan); freetdmLibCallId = FtdmCallId(callerData->call_id); } and that was the trigger to my post. Please, is the comment from the freetdm.h about the call_id within the caller_data valid? Is the zero also valid call_id? Thank you in advance Juraj

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Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 67, Issue 10
by Inge Kristiansen 30 Jan '12

30 Jan '12

Thank you, Ken. That solved my problem. Awesome. Thank you. 2012/1/30 <freeswitch-dev-request(a)lists.freeswitch.org> > Send FreeSWITCH-dev mailing list submissions to > freeswitch-dev(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > or, via email, send a message with subject or body 'help' to > freeswitch-dev-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-dev-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-dev digest..." > > Today's Topics: > > 1. Re: msrp support (Seven Du) > 2. Community Alert: Our Friend Brian West Is In The Hospital > (Michael Collins) > 3. Curl Problem (Inge Kristiansen) > 4. Re: Curl Problem (Ken Rice) > > > ---------- Videresendt e-post ---------- > From: Seven Du <dujinfang(a)gmail.com> > To: freeswitch-dev(a)lists.freeswitch.org > Cc: > Date: Mon, 30 Jan 2012 20:41:45 +0800 > Subject: Re: [Freeswitch-dev] msrp support > UPDATE on MSRP > > As the lib is GPL and incomplete and it crashes when I test with blink, I > thought it might be easier to write a simple implementation than debug the > existing code. > > So I have the following code: > > > https://github.com/seven1240/FreeSWITCH/commit/272607bf34d9748f20bd9e8082a1… > > Please take a look and give some feedback so I can make progress. > > I followed RFC 4975 with some exceptions: > > 1) MSRP in FS is always passive, so clients behind NAT can easily connect > to it > 2) MSRP service only use one port(hardcoded to 8044 for now), so it's easy > for firewall configs > 3) MSRP always works in B2BUA way, perhaps we can add bypass_media or > external MSRP relay later > 4) MSRP payload is different than RTP anyway, so I added some apps like > msrp_echo, msrp_send, msrp_recv and api like uuid_msrp_send etc. > 5) Each session use one TCP connection, no duplex > 6) Only support SDP have MSRP only, no mix of audio/videos > 7) I tried with blink on Mac and Boghe on Windows, send messages and files > with FS is sort of works with the above code. msrp_bridge is buggy but it > is possible to send messages but not files for now. > > So I have new questions for the right way to do this: > > Current code is coupled in mod_sofia and ugly, but it need stub code into > mod_sofia for SDP negotiation anyway, Ideally it should be a new mod_msrp > to decouple from mod_sofia, but mod_sofia cannot call functions in mod_msrp > I thought about the spandsp fax way but fax using RTP but MSRP using TCP so > I don't now if it's possible. Another idea would be put the msrp code into > core, the advantage would be we have another media type in core so audio > and MSRP data could be mixed in one SDP session. > > Ideas? > > Thanks. > > On Wednesday, December 7, 2011 at 10:11 AM, Seven Du wrote: > > http://jira.freeswitch.org/browse/FS-3748 > > any other comments to my other questions? I think it would be a mod_msrp > and some msrp_glue.c things. > > > Thanks. > > On Tuesday, December 6, 2011 at 11:49 PM, Brian West wrote: > > Submit this to jira.freeswitch.org and we can push things upstream. > > /b > > On Dec 6, 2011, at 12:08 AM, Seven Du wrote: > > Hi, > > We are working on msrp support in FS. With the following patch to lib > sofia, the msrp SDP can deliver to mod_sofia > > http://pastebin.freeswitch.org/17944 > > questions: > > 1) should I submit this to our jira or to the the sofia team? Any > suggestions on the patch? > > 2) As FS is a B2BUA, so I think it would be endpoint-to-endpoint bridge, > then we need start TCP/MSRP thread to handle media > > a. It should more like video handling or more like fax ? That means to > hack bridge code to also handle msrp, or should we implement api like > tx_fax/rx_fax vs tx_msrp/rxmsrp? > > b. for text over msrp, is it possible to routing the chatplan like mod_sms > > c. the first stage would be just endpoint to endpoint, shoud we implement > relay ? or even switch mode for conference? I'm very new to msrp. > > > 3) we found a msrp lib from http://confiance.sourceforge.net/ . It's GPL, > I know it's not license compatible, but as I learned from previous talk > about video, the code can put elsewhere and we just implement the logic to > hook them on. Or other lib we could use? > > > Any suggestion is appreciated. > > > Seven. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow (http://www.sparrowmailapp.com) > > _________________________________________________________________________ > 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 > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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 > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > > ---------- Videresendt e-post ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org, > freeswitch-dev(a)lists.freeswitch.org > Cc: > Date: Mon, 30 Jan 2012 09:09:41 -0800 > Subject: [Freeswitch-dev] Community Alert: Our Friend Brian West Is In The > Hospital > Hello FreeSWITCH Community, > > We wanted to let everyone know that Brian West is in St. Francis Hospital, > Tulsa, OK. He is checked into room 4325. At this point it looks like he > will be there for at least a week. > > I think it's safe to say that Brian is a dynamo not just for FreeSWITCH > but for open source VoIP and telephony. My guess is that there's not a > person who's worked with Asterisk or FreeSWITCH in the past 10 years who > hasn't been aided in one way or another by Brian. Personally I can say that > in the past five years Brian has helped me immeasurably. I think I have > literally asked him thousands of questions and he has patiently answered > them all. > > We would like to have everyone keep Brian in their thoughts and prayers. > Please feel free to send him cards and well wishes at the hospital. Also, > if you wish to send gifts and such to his home address you may do so: > > Brian West > 714 Osage Ave > McAlester, OK > 74501-6638 > > For those who are interested in helping in more practical ways, such as > helping to defray the considerable medical costs that no doubt will be > incurred, we will have more information shortly. > > Please spread the word via Facebook, Twitter, and any other medium you > have. Let's all let Brian know just how much we value him as a colleague > and friend! > > -Michael Collins > > > > ---------- Videresendt e-post ---------- > From: Inge Kristiansen <inge.kristiansen(a)gmail.com> > To: freeswitch-dev(a)lists.freeswitch.org > Cc: > Date: Mon, 30 Jan 2012 10:13:56 +0100 > Subject: [Freeswitch-dev] Curl Problem > > Hi everybody > > I have a small problem from the dial plan. I want to bill calls made by > softphones. > when I do a curl lookup from the dialplan, I can not find any value in > ${curl_response_data}. > > Testing in the cli returns correct value, so it looks correct. > > <action application="curl" data="data= > http://208.117.44.59/LCR/xxxxxxx.aspx?lega=xxxxx&legb=xxxxxx" /> > <action application="export" data="nolocal:nibble_rate=${* > curl_response_data*}" /> > > Any suggestions? > > regds > Inge > > > > > ---------- Videresendt e-post ---------- > From: Ken Rice <krice(a)freeswitch.org> > To: "freeswitch-dev(a)lists.freeswitch.org" < > freeswitch-dev(a)lists.freeswitch.org> > Cc: > Date: Mon, 30 Jan 2012 11:20:38 -0600 > Subject: Re: [Freeswitch-dev] Curl Problem > Inline... > > Keep in mind that variables in the dialplan are expanded before the first > action is even processed... If you want that variable to be usable on the > next action w/out a re-entry into the dialplan (ie: transfer) you need to > inline that application call... > > > On 1/30/12 3:13 AM, "Inge Kristiansen" <inge.kristiansen(a)gmail.com> wrote: > > > Hi everybody > > I have a small problem from the dial plan. I want to bill calls made by > softphones. > when I do a curl lookup from the dialplan, I can not find any value in > ${curl_response_data}. > > Testing in the cli returns correct value, so it looks correct. > > <action application="curl" data="data= > http://208.117.44.59/LCR/xxxxxxx.aspx?lega=xxxxx&legb=xxxxxx" /> > <action application="export" data="nolocal:nibble_rate=${* > curl_response_data*}" /> > > Any suggestions? > > regds > Inge > > > > ------------------------------ > _________________________________________________________________________ > 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 > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- //Inge Kristiansen onlineSMSC Ltd. Support : (44)2034113453 (47)21953756

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Curl Problem
by Inge Kristiansen 30 Jan '12

30 Jan '12

Hi everybody I have a small problem from the dial plan. I want to bill calls made by softphones. when I do a curl lookup from the dialplan, I can not find any value in ${curl_response_data}. Testing in the cli returns correct value, so it looks correct. <action application="curl" data="data= http://208.117.44.59/LCR/xxxxxxx.aspx?lega=xxxxx&legb=xxxxxx" /> <action application="export" data="nolocal:nibble_rate=${*curl_response_data *}" /> Any suggestions? regds Inge

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