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Codec negotiation bug?
by Juan Antonio Ibañez Santorum 16 Apr '13

16 Apr '13

Hello, I am doing some codec negotiation (early) tests but FS does not work as I expected. I can read at wiki: "When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top. If B does not accept any of the codecs, the calls fails, obviously" Codec prefs setted as: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/> FS select PCMA for A leg FS offers following SDP to B leg: v=0 o=FreeSWITCH 1365781615 1365781616 IN IP4 176.31.117.49 s=FreeSWITCH c=IN IP4 176.31.117.49 t=0 0 m=audio 19780 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 m=audio 19780 RTP/AVP 18 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Session progress from B leg contains: v=0 o=root 3311 3311 IN IP4 91.192.110.234 s=session c=IN IP4 91.192.110.234 t=0 0 m=audio 16194 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv FS codec comparison debug look as: sofia_glue.c:5094 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:30:64000] sofia_glue.c:5094 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] sofia_glue.c:3077 Set Codec sofia/external/0034950004588 G729/8000 20 ms 160 samples 8000 bits FS start comparing G729 instead PCMA selected in A leg Is that ok? is there any way to get selected PCMA for B leg without using late negotiation nor disabling transcoding Regards

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Communication between sip soft phones(X-Lite) via freeswitch
by Navnath Sonavne 16 Apr '13

16 Apr '13

Hi All, I have successfully compiled and installed freeswitch on my Linux CentOS. I have registered two softphones of X-Lite in freeswitch with extension numbers are 1100 ,1101. I have installed these softphones on two different machines of windows OS. I have created two XML dial plans for above extension numbers in usr/local/freeswitch/conf/directory/default. Now when i call 1101 from 1100 extension it did not establish a call. I know something is going wrong but what is it? I am not able to figure out. I request you all please help me to establish a call between 1100 and 1101. FYI i am reading Freeswitch 1.0.6 by Anthony Minessale , Michael S. Collins , Darren Schreiber . I am attaching 1100.xml and 1101.xml files. Regards, Navnath.

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Communication between sip soft phones(X-Lite) via freeswitch
by Navnath Sonavne 15 Apr '13

15 Apr '13

Hi, Your response : Probably your users might not be in the default context. To make call from one sip user to another both must be in same context. For testing purpose you may want to keep both of them in default context. In the user_name.xml file put context = default , this may solve your problem. My both extensions are in default context .To crosscheck it i used list_users command at fs_cli console and it shown that both phones are in default context.Here is output of list_users 1100|default|192.168.8.41|default|sofia/internal/sip:1100@192.168.9.165:29072;rinstance=ee8af9a648b2e70d||Vivekanandm|1100 1101|default|192.168.8.41|default|sofia/internal/sip:1101@192.168.8.103:26196;rinstance=a6ab3905d800b8f7||navnath|1101 As you suggested put context = default in user_name.xml but i did not find user_name.xml file in my freeswitch directory. Can you please tell me where to locate this file? Regards, Navnath.

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mod_callcenter with Active/Backup Freeswitch servers using a shared mysql database
by Ira Tessler 15 Apr '13

15 Apr '13

Configuration: Server: FS1 - Active server. All calls come into this server. All phones register to this server. Server: FS2 - Backup Server. Is online just in in case FS1 fails. If FS1 fails, all calls get routed to FS2 (by my SBC) Both FS1 & FS2 share a mysql database for core, all sofia profiles and mod_call center. The problem I am having is with mod_callcenter. When a call comes into FS1 and FS2 Freeswitch process is down, all works as expected when I call the call center application. When FS2 is online as well as FS1 it looks like FS2 is processing data in the members table(I'm guessing) and I get this log entries: 2013-04-15 13:32:40.225033 [DEBUG] mod_callcenter.c:1045 Updated Agent agent701(a)22555.sbc.cmvtesttele.com set state = Receiving 2013-04-15 13:32:40.225033 [ERR] mod_callcenter.c:1399 Member CONNECTME <+17324909007> with uuid 71b3beea-a5f2-11e2-9143-7dbd0c2f58e6 in queue mySbcQueue1(a)22555.sbc.cmvtesttele.com is gone just before we assigned an agent 2013-04-15 13:32:40.225033 [DEBUG] mod_callcenter.c:1045 Updated Agent agent701(a)22555.sbc.cmvtesttele.com set state = Waiting Since FS1 is the "live" server, I don't want FS2 to process call center calls. I want FS1 and FS2 to share the call center data in order have one copy of the agent and tier data for redundancy and so I don't have to worry about syncing multiple call center database. Before I post a Jira, I wanted to see if I am doing some wrong, or the other question does mod_callcenter have the capability or is it designed to work in this configuration? If not, anyhow have any alternative solutions? Thanks! Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira(a)connectmevoice.com

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FreeSWITCH Weekly News and Notes
by Michael Collins 15 Apr '13

15 Apr '13

Greetings! Happy tax day to those in the USA - we hope all is well with your business. Speaking of business, I thought I would relay the interesting news<http://www.cnbc.com/id/100637184>about Dish Network making a bid for Sprint. Many here in North America will be keeping a close eye on this one. Whether or not this is just a big mess<http://www.zdnet.com/softbank-sprint-clearwire-dish-figuring-out-this-merge…>or an opportunity for Sprint to become a "real" competitor to AT&T and Verizon remains to be seen. Regardless of the outcome, most of us here are hoping for a healthier Sprint so that we can avoid another duopoly. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_10>we decided to have a preliminary discussion so that we can prepare for this week <http://wiki.freeswitch.org/wiki/FS_weekly_2013_04_17>. Dave Kompel will be showing us how to build rapidly FreeSWITCH applications in MS Visual Studio 2012 and have those run under mod_managed<http://wiki.freeswitch.org/wiki/Mod_managed>. Be sure to consult this document<http://download.drknetworking.com/MeetingPreparation.pdf>so that you can get all the prerequisites installed in time for our call on Wednesday. In other news I would like to let everyone know that I spoke with Kashif Kahn over at Vestec <http://www.vestec.com>. We are gearing up for the automatic speech recognition application building contest. The winners will be announced at ClueCon 2013 <http://www.cluecon.com>. The official contest page will be posted on the ClueCon website shortly. Stay tuned for more information and be ready to start building your ASR applications! The ClueCon 2013 call for speakers recently went out and we've had a number of submissions already. We look forwarding to hearing more talk ideas, so please send those in right away. In the meantime ClueCon registration<http://www.cluecon.com/register/>is now open so be sure to get signed up, and don't forget to book your room at the Hyatt Chicago Magnificent Mile <http://www.cluecon.com/hotels/>hotel for only $169 per night. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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the user part of PAI uri
by Mickael Hubert 15 Apr '13

15 Apr '13

Hi list, I would like to retrieve the user part of PAI's uri. In fact, when I have : PAI: P-Asserted-Identity: <*tel*:+333........@freeswitch1>, my variable_sip_P-Asserted-Identity is: +333333*@freeswitch* But with: PAI: P-Asserted-Identity: <*sip*:+333........@freeswitch1>, my variable_sip_P-Asserted-Identity is: +333333 How to write only user part of PAI in my variable_sip_P-Asserted-Identity on each of the two cases ? thanks -- Cordialement Hubert Mickaël Ingénieur VOIP - Hexanet --

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SQL ERR database is locked
by mahesh katta 15 Apr '13

15 Apr '13

Hi List, """SQL ERR[database is Locked]""" When this come calls are stop. why this is come ? and how can I solve this problem , this problem occur periodically. currently I am using Freeswich-1.3XXX . Kindly suggest me how can I fix this problem. Thanking you. Best Regards, Mahesh Katta

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default timeout
by Stephen Thwaites 15 Apr '13

15 Apr '13

Hi All, Apologies for the simple question but I can't find the answer anywhere in the books, wiki, or our friend google. If somebody calls us, FS does an ORIGINATE_CANCEL after 60s but the follow-me scheme we have configured is100s. How can I increase the default call_timeout of 60s to 100s? Or maybe I am just doing something wrong! I have tried leg_timeouts, originate_timeouts both on the transfer and the bridge as well to no avail? Would be very grateful for any help or advise. Regards, Steve Some Details: - External call comes in on external profile from our voip provider. - Dialplan in the public context does a transfer to a follow-me extension 7777 in context creche-babys <action application="transfer" data="7777 XML creche-babys "/> - The 7777 extension is as follows and the call hangs up part way through the third step if nobody picks up (after 60s total). <extension name="creche-fm"> <condition field="destination_number" expression="^7777$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ringback=${nl-ring}"/> <action application="set" data="call_timeout=25"/> <action application="bridge" data="{ignore_early_media=true}user/21@${domain_name},user/20@${domain_name}"/> <action application="set" data="call_timeout=25"/> <action application="bridge" data="{ignore_early_media=true}user/22@${domain_name}"/> <action application="set" data="call_timeout=25"/> <action application="bridge" data={ignore_early_media=true}user/23@${domain_name},user/24@${domain_name},user/26@${domain_name},user/27@${domain_name},user/28@${domain_name}"/> <action application="set" data="ringback=${nl-ring}"/> <action application="set" data="call_timeout=25"/> <action application="bridge" data="{ignore_early_media=true}sofia/gateway/3120 ...voipprovidergateway/06 ...mobile number"/> </condition> </extension>

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PRI channels suspended
by Ashish gautam 15 Apr '13

15 Apr '13

Hi, I am having another problem over here. When I run "ftdm dump 1 5" from fs_cli I get my channel status as SUSPENDED. This happens for channel number 5 to 30. Also when I try to make outgoing calls more than 4 in number, four calls originate properly whereas for the rest it shows NORMAL_CIRCUIT_CONGESTION. Here is the output: span_id: 1 chan_id: 5 physical_span_id: 1 physical_chan_id: 5 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: UP type: B state: SUSPENDED last_state: RESTART txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NORMAL_UNSPECIFIED session: (none) -- States -- -- Function -- -- Location -- -- Time Offset -- DOWN => RESTART [on_dchan_up] [ftmod_libpri.c:2372] 0ms RESTART => SUSPENDED [on_timeout_t316] [ftmod_libpri.c:1952] 120008ms Time since last state change: 324790932ms Please help me getting out of this. Thanks. --Ashish

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GSMopen - huawei E1550 can't read messages
by Zohair Raza 14 Apr '13

14 Apr '13

Hi, I have a huawei e1550, it can't receive messages though sending is fine event plain message shows nothing when I send a message to the sim Here are the logs after the SIM inbox got full http://pastebin.com/kjUYHGNA Regards, Zohair Raza

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