February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Sofia and SIP Service-Route header
by Žiga Jakhel 16 Feb '11

16 Feb '11

Hi! We're having issues with a >funky< ISP - they started advertising a SIP Service-Route header pointing to nowhere (go figure), claiming they need it for their systems to function, and that our UA should not use it, if the initial address is working. Sofia, however, uses the header dilligently, sending the request to the address specified in the header (in my case going nowhere), despite the functioning base SBC address. Any ideas? Is my ISP off the rocker, or is Sofia maybe a little overzealous? Regards, Žiga Jakhel, MSc Integrum d.o.o. Smoletova 6, 1230 Domžale Slovenia, EU

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Re: [Freeswitch-users] g729 packets skipped in passthrough mode
by Javier Gallart 16 Feb '11

16 Feb '11

Hello, I've looked deeper into this issue and I might have hit a known problem: RTP_BUG_IGNORE_MARK_BIT = (1 << 2) /* A Huawei SBC has been discovered that sends the mark bit on every single RTP packet. Since this causes the RTP stack to flush it's buffers, it horribly messes up the timing on the channel. This flag will do nothing when an inbound packet contains the mark bit. */ The carrier is actually a Huawei SBC, and the problem I reported only happens when it sends the marker bit = true in all the rtp packets. I've tried <param name="auto-rtp-bugs" data="RTP_BUG_IGNORE_MARK_BIT"/> in the profile but the problem is still there, is this the right way to override this nonsense from the Huawei? Thanks Javier > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Tue, 15 Feb 2011 16:38:35 -0600 > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 56, Issue 177 > Well that's all I could think of: > > passthrough is passthrough FS never modifies anything. > you need to look harder at sip trace, pcaps or other diagnostics for > some misconfiguration. > many people use this daily. > > > On Tue, Feb 15, 2011 at 1:12 AM, Javier Gallart <jgallartm(a)gmail.com> > wrote: > > > > Anthony, thanks for the tip. I haven't seen any change though. I should > have > > mentioned that I'm focusing in the rtp stream coming form the callee to > the > > caller. The problem is that FS relays to the caller only half of the > packets > > received from the callee. The packets arrive at FS at a rate of 1 packet > > every 20 ms, and with a payload of 20 bytes each. Packets from FS to the > > caller are sent every 40 ms with a payload of 20 bytes, thus skipping > half > > of the information. > > > > Thanks > >> > >> ---------- Forwarded message ---------- > >> From: Anthony Minessale <anthony.minessale(a)gmail.com> > >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > >> Date: Mon, 14 Feb 2011 10:38:57 -0600 > >> Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough mode > >> I bet its 20ms vs 30 > >> set passthru_ptime_mismatch to true either in vars.xml or in your > >> dialplan both legs through export on the a leg or set in the a leg and > >> in the {} on b. > >> > >> > >> On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian(a)freeswitch.org> > wrote: > >> > What exactly is the problem? I see no issue here can you elaborate on > >> > what you're seeing? > >> > > >> > /b > >> > > >> > On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote: > >> > > >> >> I've tried to explicitly set ptime at 20ms at switch.conf -although > >> >> it's not necessary afaik-. Has any one experienced this same issue? > >> >> > >> >> Thanks in advance.. > >> >> > >> >> Regards > >> > > >> > > >

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connecting freeswitch to a kx-td1232
by babak yakhchali 16 Feb '11

16 Feb '11

Hi Is it possible to connect a kx-td1232 to freeswitch for connecting an office with 8 employees to the current kx-td1232 setup? and if it is possible what kind of cards do I need and how should I do that? thanx

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Persistence BLFs (or: Dear PBX please remember the BLF state)
by Patrick Plattes 16 Feb '11

16 Feb '11

Hi List :-), I'm currently switching from Asterisk to FreeSWITCH. It's really hard work for an Asterisk user, but using Asterisk becomes more and more painful even for small installations (less than 100 sip users). I know FreeSWITCH is not a drop-in replacement for the Asterisk PBX, but I don't want to change the behaviour of the PBX for the users. It's a preconception in Germany that the American people like (especially for the X-mas time) kitsch. Everyone here know the American houses with a hole bunch of blinking lights. But those decorated hoses are nothing against our offices! Our phones have up to 136 lights (BLFs). You often have to wear sunglasses at the office ;-) A typical usage of BLFs is to check if an agent is a member of the queue. I've build a simple extension to add and delete a member. I can user "presence in" and "presence out" to enable or disable the BLF, but there is one big issue. After a reboot of the phone the the user is still a member of the queue, but the BLF is off. We are using hints at Asterisk to show the user if he is a member and it works even after a reboot - with "presence" at FreeSWITCH it works (of cause) not. Does anyone have an idea how to implement it? My current extension is just for testing and so I use mod_fifo. It shouldn't be a problem to use mod_callcenter. The phone calls "queue-the_name_of_the_queue-the_name_of_the_user@pbxdomain" eg. "queue-sales_de-1000(a)freeswitch.cust" So how implement BLF persistence? Thanks, Patrick Colourized version: http://pastebin.com/ufJ7U930 <extension name="Queue Management"> <condition field="destination_number" expression="^queue-(\w+)-(\d+)$" break="on-false"> <action application="set" data="queue_name=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$1)}" inline="true"/> <action application="set" data="queue_user=${regex(${destination_number}|^queue-(\w+)-(\d+)$|$2)}" inline="true"/> </condition> <condition field="${fifo(list ${queue_name}@${domain_name})}" expression="${queue_user}"> <action application="answer"/> <action application="set" data="result=${fifo_member(del ${queue_name}@${domain_name} {fifo_member_wait=nowait}user/${queue_user} )}"/> <action application="log" data="INFO Delete FIFO agent result: ${result}"/> <action application="log" data="INFO User Logout: ${user_name}"/> <action application="sleep" data="600"/> <action application="playback" data="ivr/ivr-you_are_now_logged_out.wav"/> <action application="set" data="result=${presence(out queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/> <action application="log" data="INFO Presence OUT ${result}"/> <action application="hangup"/> <anti-action application="answer"/> <anti-action application="set" data="result=${fifo_member(add ${queue_name}@${domain_name} {fifo_member_wait=nowait}user/${queue_user} )}"/> <anti-action application="log" data="INFO Add FIFO agent result: ${result}"/> <anti-action application="log" data="INFO User Login: ${user_name}"/> <anti-action application="sleep" data="600"/> <anti-action application="playback" data="ivr/ivr-you_are_now_logged_in.wav"/> <anti-action application="set" data="result=${presence(in queue-${queue_name}-${queue_user}@${domain_name} inuse inuse)}"/> <anti-action application="log" data="INFO Presence IN ${result}"/> <anti-action application="hangup"/> </condition> </extension>

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Mod_callcenter reload xml error
by asilva＠wirelessmundi.com 16 Feb '11

16 Feb '11

Hi, Is it possible to reload mod_callcenter to force it to reload / reset the configuration form the XML and synchronized it to callcenter database. I'm using the latest git and i couldn't find any option that allows me to do that... My idea is to add / or remove queues or even change some parameters, or add more agents to the xml and with a single command reload the new configuration. I notice that i have callcenter_config queue load | unload callcenter_config agent add | del is very painful to add or remove one element by one when i remove it form the xml. I tried to reload the mod_callcenter: # reload mod_callcenter R: +OK module unloaded +OK Reloading XML +OK module loaded [...] 2011-02-16 01:34:38.155841 [DEBUG] mod_callcenter.c:1045 Updated Agent cc1_queue1_agent1 set contact = [call_timeout=30]loopback/101/local 2011-02-16 01:34:38.156843 [DEBUG] mod_callcenter.c:1045 Updated Agent cc1_queue1_agent1 set status = Available 2011-02-16 01:34:38.157845 [DEBUG] mod_callcenter.c:1045 Updated Agent cc1_queue1_agent1 set wrap_up_time = 40 2011-02-16 01:34:38.159849 [DEBUG] mod_callcenter.c:1045 Updated Agent cc1_queue1_agent1 set type = callback 2011-02-16 01:34:38.161865 [DEBUG] mod_callcenter.c:1166 Updated tier: Agent cc1_queue1_agent1 in Queue cc1_queue1 set level = 1 2011-02-16 01:34:38.163868 [DEBUG] mod_callcenter.c:1166 Updated tier: Agent cc1_queue1_agent1 in Queue cc1_queue1 set position = 5 Seams to work, in the messages i see only the agents that are defined in the XML, but when i do: # callcenter_config agent list (i still have the previous agents that where defined) freeswitch@internal> callcenter_config agent list name|system|uuid|type|contact|status|state|max_no_answer|wrap_up_time|reject_delay_time|busy_delay_time|last_bridge_start|last_bridge_end|last_offered_call|last_status_change|no_answer_count|calls_answered|talk_time|ready_time cc1_queue1_agent1|single_box||callback|[call_timeout=30]loopback/101/local|Available|Waiting|0|40|0|0|0|0|0|1297815798|0|0|0|0 cc1_queue1_agent2|single_box||callback|[call_timeout=30]loopback/105/local|Available|Waiting|0|40|0|0|0|0|0|1297815798|0|0|0|0 +OK Is it a bug in the CLI? Can anyone point me how to reload the XML configuration for the callcenter without doing it manually queue by queue, agent by agent, or tier by tier? Thanks, António Silva

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ESL Outbound: how to bridge media?
by Stephen Wilde 16 Feb '11

16 Feb '11

The problem I have is to propagate the audio coming from an "originated" outbound session to the inbound session when the outbound is in the PROGRESS MEDIA phase. When my application receives the "CHANNEL_PROGRESS_MEDIA" event from outbound session I can do a "pre_answer" on inbound session but I'm not capable to do an audio bridge. I have tried with "uuid_bridge <outbound_uuid> <inbound_uuid>" with no result probably because this api requires that at least one session must be answered. I don't want to answer to the inbound session to propagate the outbound progressing media but I want to answer to inbound only on outbound answer. Any way to do that? Stephen

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 56, Issue 177
by Javier Gallart 15 Feb '11

15 Feb '11

Anthony, thanks for the tip. I haven't seen any change though. I should have mentioned that I'm focusing in the rtp stream coming form the callee to the caller. The problem is that FS relays to the caller only half of the packets received from the callee. The packets arrive at FS at a rate of 1 packet every 20 ms, and with a payload of 20 bytes each. Packets from FS to the caller are sent every 40 ms with a payload of 20 bytes, thus skipping half of the information. Thanks > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 14 Feb 2011 10:38:57 -0600 > Subject: Re: [Freeswitch-users] g729 packets skipped in passthrough mode > I bet its 20ms vs 30 > set passthru_ptime_mismatch to true either in vars.xml or in your > dialplan both legs through export on the a leg or set in the a leg and > in the {} on b. > > > On Mon, Feb 14, 2011 at 10:28 AM, Brian West <brian(a)freeswitch.org> wrote: > > What exactly is the problem? I see no issue here can you elaborate on > what you're seeing? > > > > /b > > > > On Feb 14, 2011, at 4:48 AM, Javier Gallart wrote: > > > >> I've tried to explicitly set ptime at 20ms at switch.conf -although it's > not necessary afaik-. Has any one experienced this same issue? > >> > >> Thanks in advance.. > >> > >> Regards > > > > > > _______________________________________________ > > 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 > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > > > >

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Getting SIP trace on user level
by Aloysius Lloyd 15 Feb '11

15 Feb '11

Hi All, Is there any console command available to get SIP trance on user level . Equivalent to Asterisk "sip set debug peer <peername>" TIA Lloyd

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Fail2ban for Windows?
by Jack Loranger 15 Feb '11

15 Feb '11

Is there a Fail2ban type solution for windows? Thanks, Jack

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Alarm Conditions
by Tim King 15 Feb '11

15 Feb '11

I am looking for a way to gather the information in the following list so I may utilize it to set alarm conditions. Can someone please let me know a good method for gathering this information or if perhaps there is a newer module for reporting alarms that I am not aware of. Thanks - Tim · CDR Failed to write/send · Call Failure · H323 Gatekeeper registration error · Number of Active calls · Active Calls based on trunk · No route for call · SIP Error Codec Translation error

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