- FreeSWITCH-users - FreeSWITCH Mailing Lists

Linksys PAP2T-NA
by Sean Devoy 23 Apr '13

23 Apr '13

HI, I have a friend in Singapore trying to attach to my FS in the States. I got him registered! But when I try to call him (or vice versa) I get CODEC trouble: 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-32:2:8000:30:0]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-32:2:8000:30:0]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-32:2:8000:30:0]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-32:2:8000:30:0]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G723:4:8000:30:6300]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G723:4:8000:30:6300]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G723:4:8000:30:6300]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G723:4:8000:30:6300]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMA:8:8000:30:64000]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMA:8:8000:30:64000]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMA:8:8000:30:64000]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [PCMA:8:8000:30:64000]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G729a:18:8000:30:8000]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G729a:18:8000:30:8000]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G729a:18:8000:30:8000]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G729a:18:8000:30:8000]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-40:96:8000:30:0]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-40:96:8000:30:0]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-40:96:8000:30:0]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-40:96:8000:30:0]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-24:97:8000:30:0]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-24:97:8000:30:0]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-24:97:8000:30:0]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-24:97:8000:30:0]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-16:98:8000:30:0]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-16:98:8000:30:0]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-16:98:8000:30:0]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [G726-16:98:8000:30:0]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [NSE:100:8000:30:0]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [NSE:100:8000:30:0]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [NSE:100:8000:30:0]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [NSE:100:8000:30:0]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [telephone-event:101:8000:30:0]/[G722:9:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [telephone-event:101:8000:30:0]/[PCMU:0:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2597 Audio Codec Compare [telephone-event:101:8000:30:0]/[GSM:3:8000:20:13200] 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2671 Substituting codec PCMA@30i@8000h 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:1617 No audio codec available 2013-04-22 23:26:09.897853 [DEBUG] switch_core_media.c:2759 Set 2833 dtmf send/recv payload to 101 2013-04-22 23:26:09.897853 [NOTICE] sofia.c:5994 Hangup sofia/external/720(a)fs_bfis.bizfocused.com [CS_NEW] [INCOMPATIBLE_DESTINATION] How do I change the Linksys ATA from [PCMU:0:8000:30:64000] to [PCMU:0:8000:20:64000] So it finds a match. Thanks, Sean

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SwitchPi - FreeSWITCH Meets the Raspberry Pi with some small Open Hardware Addons
by Ken Rice 22 Apr '13

22 Apr '13

Hey Guys, May of you know I have been working to get FreeSWITCH crammed on to the RaspberryPi. And a few of you have noticed where I have been blogging about it at http://www.switchpi.org/ well its come time to ask for a little help... If you have an EE background and have recently sent boards sent off to a short run proto fab and want to assist in a little refinement of the PCB, I think its time to start heading down that road. Please contact me off list with how you can help. Where am I currently? A) FreeSWITCH Building cleaning on Raspbian (I¹ll be making an image with FS precompiled and installed shortly), B) breadboard hardware working with Python and limited C sources. Eagle Files and Source code is on Github (Linked off the blog for easy reference) K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org http://www.switchpi.org irc.freenode.net #freeswitch

2 3

Multi-tenat PBX using xml_curl
by Chris B. Ware 22 Apr '13

22 Apr '13

Hi guys, I've looked for a help on this issue on irc and mailing-list archive without success. I need to configure a multitenant PBX using xml_curl. I've followed wiki to do it. I've two domains: DomainA.voip.com DomainB.voip.com. I can register extensions to those domains but when I call one extensions from another I receive an "INVALID PROFILE" error. The only way I've found to fix this issue is to add those domains as aliases under internal profile. I thought they were automatically added: what should I set to generate their aliases automatically? Please, help. Thanks.

2 1

ERROR: switch_rtp.c:389 Failed DTMF payload check.
by Sean Devoy 22 Apr '13

22 Apr '13

HI, I submitted a jira because people get yelled at for not sending a jira on ERRORs. However, I really think this might be a Cisco SPA504G setup issue. I get "[Err] switch_rtp.c:389 Failed DTMF payload check" when I dial OUT and hit an IVR where I have to enter DTMF numbers. My latest was entering CC number, the response I got was "You entered 4", the first of 16 digits I typed! Cisco has some parameters for DTMF and my are set thusly: DTMF Playback Level: 0 DTMF Playback Length: .1 and Inband DTMF Boost: 6dB. Could those values be incorrect or hindering the process? Thanks, Sean

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Can FreeSwitch be used as a Push-to-talk server?
by MrX 22 Apr '13

22 Apr '13

Hello, I have downloaded freeswitch and read through the wiki document, as far as I know FS can behave as a SIP server which enables both call peers speak and listen at the same time. But I wonder can it act as a push-to-talk server that only one peer speak, the others listen, and use a push button to request for speaking? If I can do this, where to start(may be do some configuration or coding)? My best regards,David

3 2

H.264 high profile support
by Jeff Morriss 22 Apr '13

22 Apr '13

Hi all, currently I'm developing a SIP client for Android which is base on the PortSIP VoIP SDK, I want to setup a server in our Lab for test purpose, does freeswtich support H.264 profile? Best regards,

2 1

freeswitch 1.7.0: transport=tcp - doesn't seem to work
by Michel Brabants 22 Apr '13

22 Apr '13

Hello, I'm using freeswitch 1.7.0 and I'm trying to use transport=tcp on one side (big packets) and udp on the other side (extra data is filtered with copy_custom_headers=false). However, when the udp-side (calling side) finished call-setup with sdp, no ack on the invite 200 OK is sent to the tcp-side. Are there problems with tcp or may I be doing something wrong? The tcp-side is using a gateway which is configured with tcp. Thanks, Michel

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Test
by Blake Priddy 22 Apr '13

22 Apr '13

I haven't gotten an email since around 2 am from all of you. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

1 0

show channels question
by Siri MM 22 Apr '13

22 Apr '13

Hello, I have a scenario where I first place an incoming call into a FIFO, and then pop it out to bridge it to another leg. In a different thread in my application, I keep monitoring the channels, to ensure that my own application is in sync with whatever is happening within freeswitch. What I see is, after the call has been removed from the FIFO, and bridged, show channels still reports application as FIFO. This confuses my application, as it thinks there should be a call in the FIFO, but when it calls FIFO count, the count returned is 0. My question is, is there a way for me to differentiate, in the show channels output, if a call is indeed in a FIFO, or if it has already been popped out and bridged? One difference that I see, between the outputs of show channel before and after the bridge, is 'callee_direction' - before the call is bridged, this is generally blank, but once the call has been bridged, it is either SEND or RECV. Can I use this to differentiate between the two? Thanks!

2 1

USER_NOT_REGISTERED
by info＠shishir.com.np 22 Apr '13

22 Apr '13

hi I am using opensips as a registration server and FS as a media proxy. I dont authenticate the users on the FS but opensips does. I am using mod_curl_xml to get the user extensions and group settings. The issue I am having is when I call to the group of the users after getting data back from mod_curl_xml I get USER_NOT_REGISTERED and hangups the call. This is what I receive back from mod_curl_xml <document type="freeswitch/xml"> <section name="directory"> <domain name="c24118.c.XXXXXXX.com"> <groups> <group name="201"> <users> <user id="201"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(users/${dialed_user}@${dialed_domain})}"/> </params> </user> </users> </group> </groups> </domain> </section> </document> This is what I get on the freeswitch 2013-04-21 07:50:37.000717 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[201(a)c24118.c.XXXXXXX.com] 2013-04-21 07:50:37.000717 [NOTICE] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2013-04-21 07:50:37.000717 [DEBUG] switch_ivr_originate.c:3606 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2013-04-21 07:50:37.000717 [NOTICE] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [group] cause: [USER_NOT_REGISTERED] 2013-04-21 07:50:37.000717 [DEBUG] switch_ivr_originate.c:3606 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2013-04-21 07:50:37.000717 [INFO] mod_dptools.c:3084 Originate Failed. Cause: USER_NOT_REGISTERED Am I missing anything ? what I am doing wrong here? Thanks in advance

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