February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

vm_delete
by Darcy 05 Feb '12

05 Feb '12

I have noticed that vm_delete does not actually delete the vm file from the storage directory, it does delete the file as far as vm_list or from the endpoint, but the file stays in the directory, is this a known issue or am I missing something. Darcy

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sip_gateway dialplan variable
by Hynek Cihlar 05 Feb '12

05 Feb '12

Dear, is there a dialplan variable that would hold the gateway name an incoming call came through? Searching through the source code and wiki didn't yield any positive result. Thanks! Hynek

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Is OPenVox Gsm Card compatible for FS?
by Balamurugan Mahendran 05 Feb '12

05 Feb '12

Hi team, Can some one please let me know whether gsm card(OpenVox G400P4 QuadBand GSM Card) will work on FS, I am trying to route calls to gsm network. I saw a driver but they said its compatible only for asterisk. Do we need separate driver for this card? is there any other brand/make where I can route gsm traffic? Thanks, Bala

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It's that time of year: Buy the devs dinner!
by Michael Collins 03 Feb '12

03 Feb '12

Hello all! Each February the FreeSWITCH team converges on a small Milwaukee suburb to plot and plan how we will continue to conquer the world of open source telephony. From personal experience I can attest to the fact that it's really difficult to conquer the world on an empty stomach. :) We'd like to invite everyone to throw a few bucks into the hat to buy dinner for the FreeSWITCH core development team. The easiest way to donate is to hit the gold "donate" button on the right-hand side of the main FreeSWITCH page. Be sure to include a memo line, something like "bon appetit," so that we can keep track. Thanks again for supporting the FreeSWITCH team! We have an awesome community and you continue to show it on a daily basis. -Michael

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using public IP and Port versus sdp port
by Darcy 03 Feb '12

03 Feb '12

I have a server sitting on a public IP and all endpoints are behind routers. In some cases I get rtp where the sdp port differs from the received port. The freeswitch picks up the public ip to route to but uses the sdp port to route the rtp back to versus the public port, anyone have any suggestions. I believe I corrected this on another server awhile back but cannot remember the setting. Thanks Darcy

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extension-in-contact setting does not seem to work
by Gerry Hull 03 Feb '12

03 Feb '12

Hello, We have extension-in-contact is set to true in our gateway, but our SIP register still shows contact using the gateway name. ( sip:gw+nortelcs1k@10.98.0.50:5080) Any help is appreciated... stuck on this one. -Gerry 1. 2. <include> 3. <gateway name="nortelcs1k"> 4. <param name="username" value="83242"/> 5.  6.  7. <param name="realm" value="sands.com"/> 8. <param name="password" value="123456"/> 9.  10. <param name="extension-in-contact" value="true"/> 11. 12. <param name="proxy" value="10.98.0.37:5070"/> 13.  14. 15.  16. 17. <param name="register" value="true"/> 18.  23.  24. <param name="register-transport" value="tcp"/> 25. 26. </gateway> 27. </include> 28. 29. 30. SIP PACKET 31. REGISTER sip:sands.com;transport=udp SIP/2.0 32. 33. Via: SIP/2.0/UDP 10.98.0.50:5080;rport;branch=z9hG4bKBKK5SyQ8y5yeQ 34. 35. Max-Forwards: 70 36. 37. From: <sip:83242@sands.com;transport=udp>;tag=BKKN0U8BNX50a 38. 39. To: <sip:83242@sands.com;transport=udp> 40. 41. Call-ID: 5915f7b9-df9d-4216-a4c6-bec38807c437 42. 43. CSeq: 23809179 REGISTER 44. 45. Contact: < sip:gw+nortelcs1k@10.98.0.50:5080;tport=udp;transport=udp;gw=nortelcs1k> 46. 47. Expires: 3600 48. 49. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c2ac8f5 2012-01-27 16-04-54 -0600 50. 51. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY 52. 53. Supported: timer, precondition, path, replaces 54. 55. Content-Length: 0

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Cannot Send Display Update Warning
by Spencer Thomason 03 Feb '12

03 Feb '12

Hello, On our SBCs which consist of FreeSWITCH behind Kamailio I'm seeing the following log entry on every channel. What could be the cause and is it something to worry about? This occurs when the call receives a 183: 2012-02-03 09:53:51.966214 [WARNING] mod_sofia.c:2102 Cannot send display update to sofia/outbound/<number>@x.x.x.x Did not receive reply to last update or channel has not been answered yet. I've looked at traces and it seems that every message send is actually passed. Both Kamailio and FreeSWITCH use a single server with a public ip, just different ports. Thanks, Spencer

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Antw.: Questions regarding mod_fifo and mod_callcenter
by Weigel, Stefan 03 Feb '12

03 Feb '12

Hi, I have the same request regarding the ringtones. With mod_callcenter it's not possible to play ringing tones when acd tries to ring an agent. Am I wrong? Stefan ----- Reply message ----- Von: "Michael Collins" <msc(a)freeswitch.org> An: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Betreff: [Freeswitch-users] Questions regarding mod_fifo and mod_callcenter Datum: Fr., Feb. 3, 2012 17:17 On Fri, Feb 3, 2012 at 7:52 AM, <georg(a)riseup.net<mailto:georg@riseup.net>> wrote: > I would like to implement a quite simple queue, but I don't know if > mod_fifo is able to handle this. Would be great if someone could clarify > this: > > - One phone number, two phones > - If both or one is idle, call should be routed directly to the phone(s), > no moh should be played (just ringtones) > - if both in use, call should be routed into a queue, caller should hear > moh > - if after this a phone gets idle, call should be routed to the phone, moh > should stopped and ringtones be played to the caller > - if a call comes from the queue, and both phones are idle, both phones > should ring at the same time (I tried this with mod_fifo, is this > possible?) Isn't this possible? I'm pretty sure that this is how mod_callcenter works - like a traditional ACD. In mod_fifo it generally doesn't play ringing tones to the caller while it's looking for an agent. If anyone knows how to have mod_fifo mimic ACD by letting the caller hear ringback while dialing the agent's phone please let me know and I will add it to the wiki. -MC

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Questions regarding mod_fifo and mod_callcenter
by georg＠riseup.net 03 Feb '12

03 Feb '12

He there, I would like to implement a quite simple queue, but I don't know if mod_fifo is able to handle this. Would be great if someone could clarify this: - One phone number, two phones - If both or one is idle, call should be routed directly to the phone(s), no moh should be played (just ringtones) - if both in use, call should be routed into a queue, caller should hear moh - if after this a phone gets idle, call should be routed to the phone, moh should stopped and ringtones be played to the caller - if a call comes from the queue, and both phones are idle, both phones should ring at the same time (I tried this with mod_fifo, is this possible?) Thanks, Georg

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Inbound Calls Caller ID
by George Sapak 03 Feb '12

03 Feb '12

On my inbound calls I get incoming call ID in this format <XXXxxxxxxx;phone-context=national>, is there any way of stripping this out so that I end up with just numbers before the system starts routing the call? I have a few work arounds setup so that the call routs properly and I get the correct caller ID on my handsets, it would be nice not to have to use these. Thank You, George

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