- FreeSWITCH-users - FreeSWITCH Mailing Lists

How can I keep the A-leg from establishing early media (and playing ringback in-band) until after the B-leg answers the call?
by Jason Jeffords 31 Aug '10

31 Aug '10

My specific use case is an inbound Skype call using mod_skypopen to a SIP phone. I would like the Skype call to keep ringing as a Skype call (not a PSTN call) until the B-leg is answered. The default behaviour is the Skype call is answered by freeswitch and ringback is played over the established Skype call. Thanks in advance, Jason

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freetdm config problem
by covici＠ccs.covici.com 31 Aug '10

31 Aug '10

Hi. I was trying to config freetdm, but it complained that the name was not provided and would not load the module. Here is the freetdm.xml followed by the freetdm.conf. I have an fxs port on channel 1 and an fxo port on channel 4 -- using dahdi. Thanks for any assistance. <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="9"/>  <param name="enable-analog-option" value="call-swap"/> <param name="enable-analog-option" value="3-way"/> </settings>  <analog_spans> <span id="1">  <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="enable-callerid" value="true"/>     </span> <span id="2">  <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/>     </span> </analog_spans> </configuration> [span zt-fxs] name => FreeTDM-fxs number => 1 fxo-channel => 4 [span zt-fxo] name => FreeTDM-fxo number => 2 fxs-channel => 1 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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How to get switch_core_session for a call
by Farhan Husain 31 Aug '10

31 Aug '10

Hello, Is there a way to get the switch_core_session of a call? Is it possible by subscribing to any channel or any other event? Thanks, Farhan

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Mod Fifo
by Jonathan Pitcher 31 Aug '10

31 Aug '10

Hi all, I am attempting to setup a dynamic fifo group. I actually have the group working and can have agents log into the que and receive calls, but what I have having trouble with is Caller ID on the call to the agent from the que. I have been looking at: http://wiki.freeswitch.org/wiki/Mod_fifo And it says that to change the caller ID to the agent you can change it in 2 places, one in the script that logs the agent in and secondly in the call that gets sent to fifo in. I have choosen the second option because it allows customers to change the name of the que and have it immediately change instead of having to force users to log in and log out again to see the name change take place. When the caller dialing into the que calls the que, this is what I return to FS. <?xml version="1.0"?> <document type="freeswitch/xml"> <section name="dialplan" description="Regex/XML Dialplan"> <context name="default"> <extension name="SoapExtension"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="domain_name=1000076.pprd.nuvio.net" /> <action application="set" data="origination_caller_id_name=sam" /> <action application="set" data="fifo_music=$${hold_music}" /> <action application="answer" /> <action application="set" data="queue_name=${used_domain}$ {destination_number:4}" /> <action application="playback" data="ivr/ivr- hold_connect_call.wav" /> <action application="fifo" data="ACD121(a)1000076.pprd.nuvio.net in" /> </condition> </extension> </context> </section> </document> This is what fifo list shows : <fifo_report> <fifo name="ACD121(a)1000076.pprd.nuvio.net" consumer_count="0" caller_count="0" waiting_count="0" importance="0" outbound_per_cycle="0" outbound_priority="5" outbound_strategy="ringall"> <outbound> <member simo="1" use_count="0" timeout="60" lag="5" outbound- call-count="3" outbound-fail-count="0" taking-calls="1" status="" outbound-call-total-count="3" outbound-fail-total-count="2" logged-on- since="2010-08-30 08:02:50" manual-calls-out-count="0" manual-calls-in- count="1" manual-calls-out-total-count="0" manual-calls-in-total- count="1" ring-count="0" start-time="2010-08-30 10:08:42" stop- time="2010-08-30 10:08:42" next-available="2010-08-30 10:08:48">{fd=1000076.pprd.nuvio.net,}user/638(a)1000076.pprd.nuvio.net</ member> </outbound> <callers></callers> <consumers></consumers> <bridges></bridges> </fifo> <fifo name="cool_fifo(a)208.77.12.21" consumer_count="0" caller_count="0" waiting_count="0" importance="0" outbound_per_cycle="1" outbound_priority="5" outbound_strategy="ringall"> <outbound></outbound> <callers></callers> <consumers></consumers> <bridges></bridges> </fifo> <fifo name="manual_calls" consumer_count="0" caller_count="0" waiting_count="0" importance="0" outbound_per_cycle="0" outbound_priority="5" outbound_strategy="ringall"> <outbound> <member simo="1" use_count="0" timeout="60" lag="5" outbound- call-count="3" outbound-fail-count="0" taking-calls="1" status="" outbound-call-total-count="3" outbound-fail-total-count="2" logged-on- since="2010-08-30 08:02:50" manual-calls-out-count="0" manual-calls-in- count="1" manual-calls-out-total-count="0" manual-calls-in-total- count="1" next-available="2010-08-30 10:08:48">{fd=1000076.pprd.nuvio.net,}user/638(a)1000076.pprd.nuvio.net</ member> </outbound> <callers></callers> <consumers></consumers> <bridges></bridges> </fifo> </fifo_report> Now everything works like it should but I can't seem to get the origination_caller_id_name variable to be used in the incoming dialplan. What am I missing ? Thanks in advance, Jonathan Pitcher

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WANTED: Cookbook recipes
by Michael Collins 31 Aug '10

31 Aug '10

Hello all! We are investigating the possibility of writing a cookbook for FreeSWITCH. We would like to invite the community at large to submit ideas for recipes. I have started a wiki page here: http://wiki.freeswitch.org/wiki/Cookbook Please add your cookbook wishlist items here. If you have an actual recipe that you'd like to contribute then mention that fact as well. We are not looking for the actual recipes themselves but rather the ideas for the recipes. I've added a number of sections on the wiki but feel free to add another one if I have missed a category of recipes. Any questions please let me know! Thanks, MC

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recvEvent in ruby
by Nigel Kent 31 Aug '10

31 Aug '10

I have this code below in which I want to receive and extract dtmf events. It seems all previous events are queued and received together at this point as the while loop is run. I don't know why it does this I only need the current events to be received. Another issue happening at this point when I press any digit 0-9 on the phone it is not getting an incoming dtmf events while @con.connected #debug puts "start loop" #no digits if dtmfbuffer.length == 0 e = @con.recvEventTimed(response_timeout) # there are digits elsif dtmfbuffer.length > 0 e = @con.recvEventTimed(interdigit_timeout) end if e name = e.getHeader("Event-Name") puts "#{name}" break if name == "SERVER_DISCONNECTED" if name == "DTMF" digit = e.getHeader("DTMF-Digit") duration = e.getHeader("DTMF-Duration") dtmfbuffer << digit puts "*** Dtmf #{digit} dur:#{duration} from #{@cli}" end end if dtmfbuffer.length > 9 break end #debug puts "end loop" end

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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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directory user without password
by François Delawarde 30 Aug '10

30 Aug '10

Hello, I don't really understand the logic when no password is specified for a specific user. Is it normal that freeswitch still sends an authentication request (401 or similar) in that case? Shouldn't it directly accept and send "200 OK" on REGISTER and INVITE requests when a userid is detected to have no configured password? Thanks, François.

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iPhone SIP Client for FreeSwitch ?
by Dave Stevenson 30 Aug '10

30 Aug '10

Hi, I have an iPhone 4 (not Jailbroken), and now that IPhone OS4 supports multi-tasking, I want to use it with FreeSwitch. I'd be interested to hear what experience others have with iPhone SIP apps connecting to FreeSwitch. There seems to be a few Apps out there which would work, some free, some not. I have one which seems to work (Media5 VOIP) and supports mutli-tasking, i.e., I can have it in the background and it alerts me what a SIP call comes in, so it might be all that I need. I would just like to hear if anyone thinks that there's something better out there- which is not too expensive obviously :-) regards Dave

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Re: [Freeswitch-users] NAT traversal questions - (long)...
by Dave Redmore 30 Aug '10

30 Aug '10

Thanks for the thoughtful response. I think that it has been by some happy accident that I have not run into more NAT issues. I usually expect to have problems and then rarely do. For example, I have a FreePBX server sitting behind the pfSense as well (on a different subnet) which is my main office phone system, it is connected to my ITSP (Teliax) with absolutely no problems - no port forward or static port NATing needed - as soon as I cut over to the pfSense, I was up and running. I think you are right re: with the rport issue. At the end of the day, I think the combination of Freeswitch not detecting NAT and the HT-287 not supporting rport was what caused the problem. Will be doing some more investigating on all this - in particular I am now curious to look into how/why my RTP streams work. I have about a dozen customers running various combination's of FreePBX/ATAs/Etc and have never had audio issues - again, probably a happy accident, but now I am motivated to understand the dance that is happening between everything. Thanks again, Dave ----- Original Message ----- From: "David Ponzone" <david.ponzone(a)ipeva.fr> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Sunday, August 29, 2010 5:06:36 PM GMT -06:00 US/Canada Central Subject: Re: [Freeswitch-users] NAT traversal questions - (long)... Dave, I misread your mail the first time and did not see you sent traces. I think there are some interesting things in those. For the packet coming from your HT-287 without the static port NAT: Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 -> rport is missing For the packet coming from the one behind a DD-WRT: rport is missing too, but the source port in the Via matches the source port of the packet, so it works, the same way it works with your pfSense if you add the static port NAT. But why FS manages to guess it's behind NAT eludes me, but the NAT detecting algo in FS is clearly complex. For the packet coming from the HT-503: I think you made a mistake. You said at the beginning of your mail that it is behind DD-WRT, but before the trace, you say "one more packet coming from Comcast/SMC". Anyway, this one is interesting. rport is there, so HT-503 is rport-capable (but HT-287 is not). I would check if a configuration or a firmware upgrade could enable rport on the HT-287. Also, you wonder if modern routers have some automatic static NAT. Actually, no, but what they do quite often is to not change the source port of the packet if this port is available on the external interface. For instance, if your device sends a packet from port 5060, and this port is free on the external side of the router, it will preserve 5060. Then if a second device sends a packet from port 5060, as it is already used, it will use a random port. You end up having a pseudo static NAT behaviour for the first device on your network. I saw that on business routers from Draytek, Funkwerk and others. So perhaps your ipcop was doing that ? Wild guess: if you add a second phone with the same source port 5060 behind a DD-WRT router, I am pretty sure you will have issues with its registration. About your question "is FreeSWITCH not tagging the device as behind nat because it is on the same subnet as pfSense ?". That's quite possible. I think there are places in FS conf where you define what is local. I think it's the special keyword localnet.auto. But I really think the end of all isues is rport. If rport is not available on your device, you can still force rport on FS side: <param name="NDLB-force-rport" value="true"/> David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IP eva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 29/08/2010 à 09:01, Dave Redmore a écrit : Hello All, I ran into an issue today that has burned up most of my day troubleshooting. I have resolved the problem, but would really like to understand what caused it, or some of the internal Freeswitch plumbing that is at play so that I can learn something from all of this time I have invested. I have a Freeswitch server running that acts as a proxy to an account with an ITSP for doing T38 faxing. The Freeswitch server has a public IP address - there are four "users" who register simple FXS ATAs to my server and it then proxies to the ITSP using the "proxy_media" functionality. It has been working very well for the last 6 months or so. I have never had to deal with any NAT traversal issues - I just point the ATA to the IP to register and everything is great. Here is what the four users "looked" like - User1 : Grandstream HT-287 -> DD-WRT Router (NAT) -> Internet -> Freeswitch Proxy User2 : Grandstream HT-503 -> DD-WRT Router (NAT) -> Internet -> Freeswitch Proxy User3 : Grandstream HT-502 -> Comcast/SMC Router (NAT) -> Internet -> Freeswitch Proxy User4 : Grandstream HT-287 -> IPCOP 1.4.11 (NAT) -> Comcast Gateway -> Freeswitch Proxy (User4 is my office, so the IPCOP firewall and the Freeswitch Proxy sit on the same Comcast Gateway) As I said, this all worked perfectly without any need to "fiddle" with anything on any firewalls - worked right out of the box. So, today I changed out my IPCOP firewall for a pfsense firewall - and my HT-287 would no longer register. After much head-scratching, packet captures, etc. I found that I needed to set up a Static Port NAT for the port the HT-287 was using (5062) in order to get this to work. So, I see WHAT is happening, but I really want to know WHY it is happening. Here are the gory details: The sofia status of the profile looks like this - when the I have the Static Port NAT in place (details changed for security): _______________________________________________________________ Call-ID: 0e551b3c694a793c(a)192.168.1.137 User: 8885554525(a)173.11.22.111 Contact: "user" < sip:8885554525@192.168.1.137;fs_nat=yes;fs_path=sip%3A8885554525%40173.22.22.55%3A5060 > Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:17:03) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 173.22.22.55 Port: 5060 Auth-User: 8885554525 Auth-Realm: 173.11.22.111 MWI-Account: 8885554525(a)173.11.22.111 Call-ID: 1716488819-5062-1(a)192.168.7.150 User: 8885554544(a)173.11.22.111 Contact: "user" < sip:8885554544@192.168.7.150:5062;user=phone;fs_nat=yes ; fs_path=sip%3A8885554544%4098.255.0.11%3A5062%3Buser%3Dphone> Agent: Grandstream HT-502 V1.1B 1.0.1.63 Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 01:48:35) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 98.255.0.11 Port: 5062 Auth-User: 8885554544 Auth-Realm: 173.11.22.111 MWI-Account: 8885554544(a)173.11.22.111 Call-ID: 090ee80e1a0ec9ed(a)10.8.11.149 User: 8885554549(a)173.11.22.111 Contact: "user" < sip:8885554549@10.8.11.149:5062 > Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 Status: Registered(UDP)(unknown) EXP(2010-08-29 02:00:42) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 173.11.22.99 Port: 5062 Auth-User: 8885554549 Auth-Realm: 173.11.22.111 MWI-Account: 8885554549(a)173.11.22.111 Call-ID: 1035241259-5060-1(a)10.1.10.150 User: 8885554547(a)173.11.22.111 Contact: "user" < sip:8885554547@10.1.10.150:5060;user=phone;fs_nat=yes;fs _path=sip%3A8885554547%4098.222.55.100%3A5060%3Buser%3Dphone> Agent: Grandstream HT-503 V1.1B 1.0.1.63 Status: Registered(UDP-NAT)(unknown) EXP(2010-08-29 00:15:09) Host: 173-11-22-111-illinois.hfc.comcastbusiness.net IP: 98.222.55.100 Port: 5060 Auth-User: 8885554547 Auth-Realm: 173.11.22.111 MWI-Account: 8885554547(a)173.11.22.111 ___________________________________________________________ The "User4" account is in red. The "Contact" field is substantially different and the "Status" indicates "Registered (UDP)", rather than "Registered (UDP-NAT)" as the others. When I do a packet capture on the external NIC interface (eth0) - I see the following when the HT-287 tries to register and the Static Port NAT is NOT in place: ___________________________________________________________________ Internet Protocol, Src: 173.11.22.99 (173.11.22.99), Dst: 173.11.22.111 (173.11.22.111) User Datagram Protocol, Src Port: 11521 (11521), Dst Port: 5090 (5090) Session Initiation Protocol Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 Method: REGISTER Request-URI: sip:173.11.22.111:5090 Request-URI Host Part: 173.11.22.111 Request-URI Host Port: 5090 Message Header Via: SIP/2.0/UDP 10.8.11.149:5062;branch=z9hG4bKda48f838c8689e41 Transport: UDP Sent-by Address: 10.8.11.149 Sent-by port: 5062 Branch: z9hG4bKda48f838c8689e41 From: < sip:8885554549@173.11.22.111:5090 >;tag=c8a0d452edc5ac4b SIP from address: sip:8885554549@173.11.22.111:5090 SIP tag: c8a0d452edc5ac4b To: < sip:8885554549@173.11.22.111:5090 > Contact: < sip:88855564549@10.8.11.149:5062 > Contact Binding: < sip:8885554549@10.8.11.149:5062 > Supported: replaces, timer Call-ID: aa77d777bae71be6(a)10.8.11.149 CSeq: 100 REGISTER Sequence Number: 100 Method: REGISTER Expires: 3600 User-Agent: Grandstream HT287 1.1.0.45 DevId 000b82127390 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 _______________________________________________________________ When Freeswitch replies back with a "401 Unauthorized" - asking for further Auth - it replies back to port 5062 - so the packet never comes back (pfsense is looking for a packet back on port 11521 in this case). If I put the Static Port NAT in place - all is well, because the "Source" port shows as "5062" - the rest of the packet looks pretty much the same. Now, here is a packet coming from one of the other Users - this one comes through a DD-WRT router - here we see that the Source Port is 5060 : _________________________________________________________________ Internet Protocol, Src: 173.22.22.55 (173.22.22.55), Dst: 173.11.22.111 (173.11.22.111) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) Session Initiation Protocol Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 Method: REGISTER Request-URI: sip:173.11.22.111:5090 [Resent Packet: False] Message Header Via: SIP/2.0/UDP 192.168.1.137;branch=z9hG4bK665bc67a1c64292b Transport: UDP Sent-by Address: 192.168.1.137 Branch: z9hG4bK665bc67a1c64292b From: "fax" < sip:8885554525@173.11.22.111:5090 >;tag=8dc68b35111c4261 To: < sip:8156564525@173.15.28.101:5090 > Contact: < sip:8885554525@192.168.1.137 > Contact Binding: < sip:8885554525@192.168.1.137 > Call-ID: 0e551b3c694a793c(a)192.168.1.137 CSeq: 503 REGISTER Sequence Number: 503 Method: REGISTER Expires: 3600 User-Agent: Grandstream HT287 1.1.0.45 DevId 000b821203c5 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 ______________________________________________________________________ Here is one more packet coming from a Comcast/SMC Router - again, the source port is correct: ______________________________________________________________________ Internet Protocol, Src: 98.244.55.100 (98.244.55.100), Dst: 173.11.22.111 (173.11.22.111) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5090 (5090) Session Initiation Protocol Request-Line: REGISTER sip:173.11.22.111:5090 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK58981045;rport Transport: UDP Sent-by Address: 10.1.10.150 Sent-by port: 5060 Branch: z9hG4bK58981045 RPort: rport From: < sip:8885554547@173.11.22.111:5090;user=phone >;tag=138706651 To: < sip:8885554547@173.11.22.111:5090;user=phone > Call-ID: 1035241259-5060-1(a)10.1.10.150 CSeq: 79875 REGISTER Sequence Number: 79875 Method: REGISTER Contact: < sip:8885554547@10.1.10.150:5060;user=phone >;reg-id=1;+sip.instance="< urn:uuid:00000000-0000-1000-8000-000B821F9A84 >" Contact Binding: < sip:8885554547@10.1.10.150:5060;user=phone >;reg-id=1;+sip.instance="< urn:uuid:00000000-0000-1000-8000-000B821F9A84 >" Max-Forwards: 70 User-Agent: Grandstream HT-503 V1.1B 1.0.1.63 Supported: path Expires: 300 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 ___________________________________________________________ So, here are my questions: - Why is the Sofia Status so much different for the registration coming through the pfSense firewall. It looks like it doesn't get tagged as being NAT'd and the "Contact" info is much less. - Do most modern routers automatically Static Port NAT any SIP traffic? Both DD-WRT and SMC routers appear to be doing this - and not just on a simple Port bases (UDP 5060 only), because one of these examples is on 5062. Are these "SIP aware" firewalls that are doing this automatically, as the IPCOP did before? - Is the extra "Contact" data in the last packet example different because it is a different UA (HT-503 rather than an HT-287) - Is Freeswitch not flagging the registration from my office (User4) as being NAT'd because it is coming in on the same subnet as the interface Freeswitch received the packet on (Freeswitch is at 173.11.22.111 and pfsense is at 173.11.22.99)? Sorry for this terribly long posting - I'm just very curious to understand what is going on here, now that I have collected all this information. Thanks, Dave _______________________________________________ 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 _______________________________________________ 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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