- FreeSWITCH-users - FreeSWITCH Mailing Lists

Incoming packet queue at RTP socket
by Yehavi Bourvine 04 Sep '14

04 Sep '14

Hello, I've noticed recenty that there is an input queue of 1832 bytes on some of the active RTP ports. In our case all th phones (about 400) are on the same profile and RTP is passed via FreeSwitch. Is this normal? Thanks, __Yehavi:

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PostgreSQL :(
by Chris Bardos 04 Sep '14

04 Sep '14

Hi All, I’m trying to make Freeswitch store everything in Postgres instead of using SQLite. I’ve put the dsn string in the xml sheets following these instructions: https://wiki.freeswitch.org/wiki/FreeSWITCH_databases#DSN_with_username_and… https://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core However, it is still using the SQLite databases. I’m sure there is something easy that I’m overlooking, but I can’t figure it out. Could someone please point me in the right direction? Thanks in advance! Chris Bardos

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Bridge call without hanging up more then one time
by sami sakly 03 Sep '14

03 Sep '14

Hello, Is it possible to make a second bridge after the first call ( bridge with b leg ) ended without calling again the a leg ? My debug screen shows : switch_ivr_bridge.c :1879 originatee uuid xxx-xxx-..-x is not present . In my system I need to have the a leg ( agent ) always available (call answred and have a known uuid) to bridge to him calls from customer without the need to issue call to the agent for every time. Thanks, Best Regards, S.Sakly

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The SIPTLS/SRTP B2BUA saga continues
by Tim Smith 03 Sep '14

03 Sep '14

Siptrace at http://pastebin.freeswitch.org/pastebin.php?dl=23211 To protect the innocent, the following IP mappings have been applied : 10.1.2.3 = SIP Phone (no TLS/SRTP support) 172.16.1.2 = VoIP Provider (TLS/SRTP) 192.168.1.11 = Freeswitch B2BUA The VoIP provider has had sight of this siptrace and flatly denies its a problem on their side and says its a problem on my side. There are millions of Freeswitch XML configuration files and I'd really appreciate a pointer in the right direction !

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MEDIA_TIMEOUT vs. NORMAL_TEMPORARY_FAILURE
by Oleg Stolyar 03 Sep '14

03 Sep '14

Hi guys, I use WebRTC to connect to FreeSWITCH. Sometimes I get a MEDIA_TIMEOUT in the logs which disconnects the leg. My rtp-timeout-sec is 300 and MEDIA_TIMEOUT seems to happen after 5 minutes, so it's all good. I thought that this would happen when the user closes the browser or their computer crashes or their network goes away. However, I tested these scenarios and in all of them FreeSWITCH disconnects after about 30 seconds with the message NORMAL_TEMPORARY_FAILURE. It's probably detecting that the websocket is no longer active. Again, all good. My question is what scenario would result in the MEDIA_TIMEOUT rather than NORMAL_TEMPORARY_FAILURE? For example, would it happen if the user went on mute for 5 min? Is there something else?

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Codec Error!
by Ali R. 03 Sep '14

03 Sep '14

Hello All, I'm getting the below error while only in pass-through mode (proxy_media=true) but it's fine when the disable-transcoding is set on the sofia profile. [WARNING] switch_core_codec.c:703 Codec G723 Exists but not at the desired implementation. 8000hz 20ms [ERR] switch_core_media.c:2285 Can't load codec? This error does not occur in case of another pass-through codec(G729) Here are my settings when in proxy media mode: Sofia Profile: inbound-late-negotiation = true inbound-codec-negotiation = greedy (tried scrooge) I also tried rtp-autofix-timing=true (no luck) DialPlan: proxy_media=true bridge FS version tried: 1.5.13b & 1.2 FS Logs: with proxy-media (failed) http://pastebin.freeswitch.org/23205 with disable-transcoding is set (success) http://pastebin.freeswitch.org/23206 A pcap file can be supplied off the list. Your help and suggestion is very much appreciated. Best Regards, Ali_R

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For those ubuquiti lovers.
by Blake Priddy 03 Sep '14

03 Sep '14

Would be awesome to throw csipsimple on this or if the app on there works. Ordering 2 for fun :) http://www.ubnt.com/unifi-voip/overview/ -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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RECOVERY_ON_TIMER_EXPIRE + Changing Hangup Cause
by Massimo Varriale 03 Sep '14

03 Sep '14

Hi All! This is my first question...and what question... I'm a skilled developer and I'm almost familiar with telephony in general since it's 15 years experience in telco companies. However I'm still facing a problem I couldn't find any solution yet and i hope someone could help in finding a solution As my present company can't afford an expensive soft switch I'm using Freeswitch 1.4 with a simple dialplan.xml and some scripts to load CDRs into mysql to send incoming calls to my remote peers. At present the live traffic is only 1 incoming IP and 1 outgoing peer, quite easy. On some calls I'm getting "408 Request Timeout" and FS is giving me a RECOVERY_ON_TIMER_EXPIRE cause (IDSN 102). The problem is that as most of the switches are not able to make automatic rerouting based on this cause and so, I will like to use instead the ISDN Cause 18 (I suppose "480 Temporarily unavailable"). Is this possible? I've used this snippet of code that use another relcause but with same concept (taken as is and pasted into the dialplan) but doesn't work as the 408 is given after 183. <extension name="AllDestinations_Inbound"> <condition field="destination_number" expression="^"> <condition field="network_addr" expression="^154\.48\.198\.190$"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=19"/> <action application="bridge" data="sofia/gateway/remote_outbound_peer/${destination_number}"/> <action application="transfer" data="408to503"/> </condition> </extension>  <extension name="408to503"> <condition field="${proto_specific_hangup_cause}" expression="<a href="sip:408">sip:408"> <action application="set" data="sip_ignore_remote_cause=true"/> <action application="respond" data="503"/> <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension> How to correctly remap causes? If there is no way to change normal behaviour, do you think could be possible also to change the source code and recompile FS accordly? Thank yo so much and sorry for this long post I hope someone could help Max

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Codec Error
by Ali R. 03 Sep '14

03 Sep '14

Hello All, I'm getting the below error while in pass-through mode (proxy_media=true) but I get success when the disable transcoding is set on the sofia profile. [WARNING] switch_core_codec.c:703 Codec G723 Exists but not at the desired implementation. 8000hz 20ms [ERR] switch_core_media.c:2285 Can't load codec? I did hear about broken SIP devices out there but since it works fine when disable transcoding is set, I thought I give it a shot on the mailing list and while at it, this error does not occur in case of another pass-through codec(G729) Here are my settings when in proxy media mode: Sofia Profile: inbound-late-negotiation = true inbound-codec-negotiation = greedy (tried scrooge) I also tried rtp-autofix-timing=true (no luck) DialPlan: proxy_media=true bridge FS version tried: 1.5.13b & 1.2 with proxy-media (failed) http://pastebin.freeswitch.org/23205 with disable-transcoding is set (success) http://pastebin.freeswitch.org/23206 Your help or suggestion is very much appreciated. Best Regards, Ali_R

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Force minimum conference member dinamically
by Antonio Silva 03 Sep '14

03 Sep '14

Hi, Is it possible to set the minimum members in a conference dynamically? for example: conference <confname> min 1 In my case, in doing conference with bridge, from the dialplan: <application="conference" data="bridge:....."> But this forces the minimum number of members to 2, so when one member hangup, the conference is ended. The idea is to still use the bridge conference, but once the conference is created via an external command change the minimum conference members to 1. Thanks, António

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