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Getting 403 trying to register an ATA to freeswitch
by Babak Yakhchali 26 Dec '16

26 Dec '16

Hi I have a grandstream ata(HT701) behind nat which is trying to register to freeswitch with public ip but it is getting 403. Zoiper is working fine with same configuration. These are sip messages after enabling siptrace recv 812 bytes from udp/[x.x.x.x]:5060 at 06:39:06.476367: ------------------------------------------------------------------------ REGISTER sip:y.y.y.y:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.33.109:5060;branch=z9hG4bK951798483;rport From: "100" <sip:100@y.y.y.y:5070>;tag=1551799885 To: <sip:100@y.y.y.y:5070> Call-ID: 1772924103-5060-2(a)BJC.BGI.DD.BAJ CSeq: 2012 REGISTER Contact: <sip:100@192.168.33.109:5060 >;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B829D0E35>" Authorization: Digest username="100", realm="y.y.y.y", nonce="a5163fda-cb35-11e6-bc3a-6bd3c5b5a0c6", uri="sip:y.y.y.y:5070", response="b3e394d5f59e1e3c79cc5ec596d132e1", algorithm=MD5, cnonce="09249246", qop=auth, nc=00000008 Max-Forwards: 70 User-Agent: Grandstream HT701 1.0.8.2 Supported: path Expires: 180 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 ------------------------------------------------------------------------ send 548 bytes to udp/[x.x.x.x]:5060 at 06:39:06.569125: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.33.109:5060 ;branch=z9hG4bK951798483;rport=5060;received=x.x.x.x From: "100" <sip:100@y.y.y.y:5070>;tag=1551799885 To: <sip:100@y.y.y.y:5070>;tag=0r6HXSrXZD44a Call-ID: 1772924103-5060-2(a)BJC.BGI.DD.BAJ CSeq: 2012 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.14+git~20131019T195513Z~44680b31d9 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 how can I debug this? thanks

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Set outgoing codec list in dialpeer/gateway. Possible?
by Dmitry Mordovin 25 Dec '16

25 Dec '16

Hello Googling several hours but can't find example how to configure special codec list (outgoing codecs offer) inside dialpeer or gateway config xml Guys, please help! Happy new year!!! Dmitry.

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mod_xml_curl - directory only
by Tihomir Culjaga 24 Dec '16

24 Dec '16

Hello everyone, i'd like to use mod_xml_curl for serving directory. <binding name="my_registrations"> <param name="gateway-url" value="http://192.168.5.150:8080/getuser" bindings="directory"/> <param name="method" value="GET"/> <param name="timeout" value="10"/> </binding> ...and as far as registrations ( SIP or verto ) it is ok, meaning my devices do register. i return something like this: <document type="freeswitch/xml"> <section name="directory"> <domain name="$${domain}"> <params> <param name="dial-string" value=" {^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(*/${dialed_user}@ ${dialed_domain})},${verto_contact(${dialed_user}@${dialed_domain})}"/> <param name="jsonrpc-allowed-methods" value="verto"/> </params> <groups> <group name="nexios-cc"> <users> <user id="agent1" number-alias="1031" cacheable="10000"> <params> <param name="a1-hash" value="f40e695952e177a9224ca3f54bf6ec5a"/> </params> <variables> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="EXT 1031"/> <variable name="effective_caller_id_number" value="1031"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> but when i place a call i get a request with action=user_call with no other info than user xxx is trying to place an outgoing call without saying where :=) section=directory&tag_name=domain&key_name=name&key_value=192.168.5.150&Event-Name=REQUEST_PARAMS&Core-UUID=15731076-c3d5-4ad6-8518-41b9c42140be&FreeSWITCH-Hostname=bladeranger&FreeSWITCH-Switchname=bladeranger&FreeSWITCH-IPv4=192.168.5.150&FreeSWITCH-IPv6=%3A%3A1&Event-Date-Local=2016-12-22%2009%3A44%3A56&Event-Date-GMT=Thu,%2022%20Dec%202016%2008%3A44%3A56%20GMT&Event-Date-Timestamp=1482396296153258&Event-Calling-File=mod_dptools.c&Event-Calling-Function=user_outgoing_channel&Event-Calling-Line-Number=4129&Event-Sequence=7368&as_channel=true&action=user_call&originate_timeout=0&origination_uuid=bf0fd829-56df-48c4-8112-41ba777ee91d&ignore_early_media=true&last_route=true&origination_caller_id_number=xxxxxxxx&origination_caller_id_name=xxxxxxxx&key=id&user=agent1&domain=192.168.5.150 So, here i got few questions :=) 1. purpose=null, action=sip_auth - i send user info here (SIP) 2. purpose=null, action=jsonrpc-authenticate - i send user info here (Verto) 3. purpose=null, action=user_call - what do i send here ? 4. purpose=null, reverse-auth-lookup - what do i return here ? 5. purpose=gateways - i will send "not found" 6. purpose=network-list - i will send "not found" 7. purpose=publish-vm - i will send "not found" Thanks to everyone who is going to enlight these few things above :=) regards, Tihomir.

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one-way audio bridging DTLS and non-DTLS channels
by Dave Horton 24 Dec '16

24 Dec '16

I’ve got a scenario where I am bridging two channels; one from a webrtc client established using DTLS, with a second channel established to a far-end session border controller not using DTLS. Audio from the web rtc client is heard by the far end client that is behind the SBC, but audio from the SBC is not heard by the webrtc client. I’ve got a pastebin here: http://pastebin.com/J9P63JUU I wonder if anyone can take a look and indicate whether they see any errors, or anything indicative of one-way audio? The DTLS connection itself seems to be set up fine, and I see no obvious errors myself. More generally, is there anything to keep in mind in bridging scenario of this type, or any reason why audio from the “insecure” connection could not be bridged over to the “secure” connection? Dave

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When activate MoH two times from different legs, the call is silent or terminated
by José Lopes 23 Dec '16

23 Dec '16

Hello, I have a FreeSwitch with vanilla configuration connected to an SIP Provider. I configured the proxy-hold=true configuration on Sofia Internal profile so when i activate the Hold on internal sip regs, it will be forwarded to SIP Provider. The SIP Provider provides MoH when requested. I have a problem that results on the call on silent or terminated by SIP Provider when I do the next actions: 1- I receive a call from the SIP Provider 2- the caller activate MoH and deactivate MoH 3- the callee activate the MoH 4- Now the call is silent (with other SIP Provider the call is dropped by SIP Provider) I notice that on step 3, the FreeSwitch send a Re-INVITE without SDP, so I think this causes the call to be silent, or sometimes the SIP Provider drop the call. I replicate this issue creating a sipp scenario and verified this issue. SIPP scenario to replicate this issue https://pastebin.freeswitch.org/view/afbae914 Freeswitch logs: https://pastebin.freeswitch.org/view/7f36d5ba Command to use sipp scenario sipp -i 172.21.0.1 172.21.0.11:5080 -s 1001 -sf scenario.xml -m 1 -l 1 -r 1 -rp 1000 #172.21.0.1 -> local ip #172.21.0.11 -> Freeswitch IP Best regards, Jose Lopes

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mod_conference - alone/on-hold timeout ?!
by Anonim Stefan 22 Dec '16

22 Dec '16

Hi, Is there such a thing as alone-timeout or on-hold-timeout for mod_conference which will end a conference for an alone conference user ? I could find "terminate-on-silence" conference profile parameter which I might use, but id doesn't seem to work for me (I have moh disabled and still doesn't terminate). Thank you, Stefan

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g729 license issue
by devang nathwani 22 Dec '16

22 Dec '16

We have configure G729 license and then i have upgrade Freeswitch version and trying to configure G729 license again but getting this error and it crashed Freeswitch. freeswitch@my-switch <freeswitch@GCN-VOICEMAIL>> load mod_com_g729 +OK Reloading XML +OK 2016-12-22 06:27:19.546361 [INFO] mod_com_g729.c:254 G.729A license count => 4 2016-12-22 06:27:19.546361 [CONSOLE] switch_loadable_module.c:1538 Successfully Loaded [mod_com_g729] 2016-12-22 06:27:19.546361 [CRIT] switch_loadable_module.c:197 Failed to load codec interface G.729 from mod_com_g729 due to bytes per frame 608487423 exceeding buffer size 8192. 2016-12-22 06:27:19.546361 [NOTICE] switch_loadable_module.c:338 Adding API Function 'g729_count' 2016-12-22 06:27:19.546361 [NOTICE] switch_loadable_module.c:338 Adding API Function 'g729_available' 2016-12-22 06:27:19.546361 [NOTICE] switch_loadable_module.c:338 Adding API Function 'g729_info' 2016-12-22 06:27:19.926299 [INFO] mod_enum.c:878 ENUM Reloaded 2016-12-22 06:27:20.086293 [INFO] switch_time.c:1415 Timezone reloaded 1750 definitions Can any one help me please?

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SIP ATA and dect phone suggestions
by Rajil Saraswat 22 Dec '16

22 Dec '16

Hello, I have been using a Linksys SPA-3102 ATA with a Panasonic KX-TG7120E dect phone for quite some time. However, I am looking to upgrade these with devices which support HD audio. My preference is to have a cordless device for home environment. Also a dect phone +SIP ATA seems to me a better choice over an integrated IP cordless phone since the dect phone is cheaper to replace in the long run. Can i have some suggestions on what people use? Thanks

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How can I enable TALK events for Sangoma TDM endpoints?
by D D 22 Dec '16

22 Dec '16

Hi, How can I enable TALK events for Sangoma TDM endpoints? Thanks,David

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[FS Unable to evaluate large regex]
by Bilal Abbasi 22 Dec '16

22 Dec '16

Hi users, i am getting some weird regex problem with my FS, if you have your regex length in more then 3000 characters if does not accept that in the condition, i split my regex in around 2000 characters, now i can see that FS is unable to resolve the Below Regex in the condition( i have a lot of other concatenated with that) ^(0092|\+92|92|0)313(9([0-9][0-9])|5([0-9][0-9])|0(0[59]|1[59]|2[6789]|5[01234567]|8[56789]|9[01278934])|1(2[6789]|5[01234567]|8[56789]|9[0-9]))[0-9]{4}$ But when i try that alone in the condition that works, i even tried to execute the whole in online regex tester and that did worked, but not working on FS. There must be a length of regex in condition that FS can evaluate, So i) i want to know that length so i can split my regex even shorter.and FS can evaluate that. ii) How FS people will look into this issue, so they can guide/fix this issue. Regards Abbasi

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