April 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

freeswitch.org SPF record does not PASS messages from JIRA server
by Steven Schoch 06 Apr '17

06 Apr '17

I'm not sure who manages it, but the SPF record for freeswitch.org is v=spf1 mx a ip4:209.105.235.8/28 a:lists.freeswitch.org ?all First of all, 209.105.235.8/28 is not even the way it's normally written, as the "/28" masks out the last ".8". The issue is that notifications from JIRA come from nobody(a)freeswitch.org, but the JIRA server sends from an IP address of 209.105.235.55, which is not in that address block. I suspect the correct ip4 network is 209.105.235.0/26, or possibly even 209.105.235.0/24. -- Steve

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Multiple destination with mod_httapi
by 'Yemi Obembe 06 Apr '17

06 Apr '17

I am trying to use mod_httapi to call multiple SIP endpoints from my script. Here is what the output of my script looks like: <document type="text/freeswitch-httapi"> <work> <dial context="default" Dialplan="XML">sip:username@ip</dial> <dial context="default" Dialplan="XML">sip:username2@ip</dial> </work></document> However, I notice that that the dial only gets to the second <dial> endpoint: sip:username2@ip. I suspect it might have to do with sofia because log from fs_cli shows both were processed. 2017-04-06 14:21:29.279911 [DEBUG] mod_httapi.c:1229 Process Tag: [dial]2017-04-06 14:21:29.279911 [DEBUG] switch_ivr.c:2160 (sofia/internal/number(a)xxx.com) State Change CS_EXECUTE -> CS_ROUTING2017-04-06 14:21:29.279911 [NOTICE] switch_ivr.c:2167 Transfer sofia/internal/number(a)xxx.com to XML[sip:username@ip@public]2017-04-06 14:21:29.279911 [DEBUG] mod_httapi.c:1229 Process Tag: [dial]2017-04-06 14:21:29.279911 [NOTICE] switch_ivr.c:2167 Transfer sofia/internal/number(a)xxx.com to XML[sip:username2@ip@public] Any ideas what could be wrong? Or a better way to do this? Thanks

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Sending remote video in early media or before the remote end answer the call
by Antonio Silva 06 Apr '17

06 Apr '17

Hi team, Is it possible to send video to the remote before it answer the call? What i want to accomplished is: . A (video phone) call to FS . FS answers the call . FS call B (video phone as well) ----> sending video from A (video phone) to B before it answers the call Or other option be able to redirect the rtp stream from A to B as soon we get rtp from A, this probably is not possible because we don't have a media stream open with B. Thanks. -- Saludos / Regards / Cumprimentos, António silva

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multiple sip profiles
by Rick Jarvis 06 Apr '17

06 Apr '17

Trying to get my head around setting TLS up. I have one internal sip profile, but multiple domains & dialplans. If I want to enable TLS on multiple domains, I think I’m right in saying that I will then need multiple sip profiles, in order to set the TLS up in each? I guess my confusion comes from understanding the difference between domains, profiles and sip profiles. I’m not even sure exactly what question I’m asking here, so apologies!

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mod_freetdm not included in FS 1.6 Debian Jessie package?
by Neil Patel 06 Apr '17

06 Apr '17

Hi All, I just installed production FS on Debian Jessie following wiki instructions. When I went to enable mod_freetdm I saw that mod_freetdm.so was not found. Indeed, I don't see it in /usr/lib/freeswtich/mod. The instructions on Sangoma wiki hint at building and merging their freeTDM, but the docs point to broken links. How do I get the freetdm module for FS 1.6? Thanks.

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Issue with 403 Forbidden outbound
by Rick Ewart 05 Apr '17

05 Apr '17

Hi All. I apologize for what seems to me to be a newbie question but I have been searching and beating my head against the wall. Hoping someone might be able to help as I know it has to be simple and I am just too dense to find it online in the myriad of available information. I have a SBC with a user setup on it (it's a commercial SBC that is freeswitch based). If I setup a grandstream phone and register to it, it works great and I can place an outbound call without issue. Nothing fancy in my grandstream config - hostname of SBC in the SIP server field, username in the Account Name, SIP User ID and Authenticate ID fields, and a password. Registers immediately and all is well. If I setup a freeswitch system to do the same, it doesn't want to work for me. It seems to register fine and I see it registered when I do "sofia status gateways" (and on the SBC too), but calls fail with 403 forbidden from the SBC. Profile::Gateway-Name Data State Ping Time IB Calls(F/T) OB Calls(F/T) ========================================================================== ======================= external::IsleCall sip:ewart123@mysiphost.ewart.net REGED 0.00 0/0 2/2 Doing a packet capture at the SBC of the working and not working calls, the key difference appears to be the From field. On the failing one I see: From: "1234 Rick" <sip:1234@99.122.123.123>;tag=raKDjmyU0vNfE While on the working one I see: From: "ewart123" <sip:ewart123@mysiphost.ewart.net>;tag=263774504 My outbound gateway configuration on the freeswitch is: <include> <gateway name="IsleCall"> <param name="proxy" value="mysiphost.ewart.net"/> <param name="username" value="ewart123"/> <param name="password" value="somepassword"/> <param name="realm" value="mysiphost.ewart.net"/> <param name="extension" value="ewart123(a)mysiphost.ewart.net"/> <param name="caller-id-in-from" value="true"/> <param name="register" value="true"/> <param name="expire-seconds" value="600"/> <param name="retry-seconds" value="30"/> </gateway> </include> Seems to me I am missing something simple to tell it to not use my internal extension and IP but instead reflect the registered username at the SBC, but I will be darned if I can figure out what that should be. Thanks in advance for any assistance you may be able to provide. Rick Ewart

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MD5 hash calculation on gateway
by Gregor Nanger 05 Apr '17

05 Apr '17

Can someone point me where in source code is MD5 has calculation for gateways. I want to store provider password as a1-hash string and respond with this string instead of calculating hash wiht user:domain:password, because do not want to store password in plain string in xml file. Best regards, Gregor

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a1 password gateway
by Gregor Nanger 05 Apr '17

05 Apr '17

Can I use a1-hash parameter in gateways configuration? Provider request to send hashed password they sent instead. Best regards, Gregor

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Re: [Freeswitch-users] Memory leakage issue
by Siju Nair 05 Apr '17

05 Apr '17

Hi team, Please help me on below issue !! Sent from my iPhone > On 09-Mar-2017, at 4:44 PM, Siju Nair <siju.irs(a)gmail.com> wrote: > > Hi team, > > Please help on below issue ! Also my CPU avg is 42 on calls ! > > Sent from my iPhone > >> On 08-Mar-2017, at 7:53 PM, Siju Nair <siju.irs(a)gmail.com> wrote: >> >> Please check the attached screen shot for more clarification >> >>> On Wed, Mar 8, 2017 at 7:48 PM, Siju Nair <siju.irs(a)gmail.com> wrote: >>> Hi team, >>> >>> My FS server got hung, under server logs due freeswitch server got >>> hung up. Below is the log i found.. >>> >>> echo 0 > /proc/sys/kernal/hung_task_timeout_secs"disables this message >>> info: task freeswitch: 3034 blocked for more than 120 secs. >>> >>> and then >>> freeswitch > waiting for background process to be ready. >>> >>> Please help me on the same. >>> >>> Sent from my iPhone >> >> <ccmmydqld_hang.PNG>

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Issue in handling 300 Multiple choices
by John Nash 05 Apr '17

05 Apr '17

I am sending call to a Redirect server (Routing) I have a context named "redirected" in default.xml When I receive an Invite I send it to redirect server and receive 300 Multiple Choices from redirect server, but this redirect is not being transfered to "redirected" context rather freeswitch tries to send Invite as per contact in 300 response. I also have <param name="manual-redirect" value="true"/> in my sip profiles I dont know why it behaves in such a way. I am using freeswitch 1.6

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