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Alert-Info and Polycom SIP v4
by Spencer Thomason 19 Mar '13

19 Mar '13

Hello all, I'm attempting to setup distinctive ring on a few Polycom IP-650/670s running SIP firmware v4. It seems they want a "non-compliant" Alert-Info header to determine the ring class. See: http://community.polycom.com/t5/VoIP/Paging-i-e-auto-answer-not-working-aft… Is there a way to add a "raw" Alert-Info header? i.e. without brackets? I attempted to set the alert_info variable and sip_h_Alert-Info with no luck. An example of what I'm trying to achieve is: Alert-Info: info=loud-ring Thanks, Spencer

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tracking a call across multiple servers
by Jason Holden 19 Mar '13

19 Mar '13

Any suggestions for adding a unique identifier to a call to be able to track it across multiple servers when tracing a call path?

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g729 and AMR codec requirements
by Abdul Basit 19 Mar '13

19 Mar '13

Hi, I am planing to setup FS based conferencing solutions with 500 dynamic conference rooms with 3 ~ 5 participants in each room. Need suggestions on hardware dimensioning and codec requirements. I am thinking on using AMR and G729 codecs. How many codec licenses are required? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445

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MWI notifications about new voicemails is not updating
by Yuriy Nasida 19 Mar '13

19 Mar '13

Hi guys. I use 2 internal sip profiles with 5060\5061 sip ports.Both sip profiles has has same settings exclude sip ports and alias (they just has different sip ports too)Domain name = IP address. Do you know why MWI notifications about new voicemails is not updating for registered sip devices via one sip profile in case of simultaneous using of these two sip profiles ? I.e. all works fine but with 5060 or 5061 sip profile only. Not for both of them simultaneously. Thanks.

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Api originate call drop
by ashish gautam 19 Mar '13

19 Mar '13

Hi, I am generating an outbound call by connecting to event socket using "api originate command". I am generating call to PSTN netwrok via freetdm module. Now the problem is as soon as I the call is generated and reaches the destination, it immediately drops and gets disconnected. The Api command I am sending to the socket is "api originate freetdm/1/a/8802865008 47673501 XML public" The dialplan in the public directory (named as 01_DID.xml) is : <include> <extension name="public_did"> <condition field="destination_number" expression="^(47673501)$"> <action application="python" data="test"/> </condition> </extension> </include> I have installed mod_python and when I make an incoming call to this extension. It works as expected. -- REGARDS ============================================ *Ashish Gautam*

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Weird ongoing RTP flow
by David Villasmil 19 Mar '13

19 Mar '13

Hello Guys, I have a weird case in which we made a test call from 1.2.3.4 to x.y.z.a and hung up after 10 seconds, everything went fine, CDRs match on both sides, etc. all good... But when I did an ngrep i'm still seeing rtp flowing... anyone ever see this before??? I should point out that this server is doing thousands of calls per hour, no problems whatsoever... The ngrep command is: "ngrep -sip host 1.2.3.4" Thanks! # U 1.2.3.4:20622 -> x.y.z.a:12598 ...d7BU.a.'.x.@.......xz........ # U x.y.z.a:12598 -> 1.2.3.4:20622 .......... Xx.........x.@....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...e7BV|a.'.x.........x......... # U x.y.z.a:12598 -> 1.2.3.4:20622 .......".. Xx.........x.@....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...f7BW.a.'.xR........x.@....... # U x.y.z.a:12598 -> 1.2.3.4:20622 .......... Xx>........xS@....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...g7BW.a.'.xz........x......... # U x.y.z.a:12598 -> 1.2.3.4:20622 .......b.. Xx.........x.@....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...h7BX\a.'.x.........xR........ # U x.y.z.a:12598 -> 1.2.3.4:20622 .......... Xx~........x.@....... # U 1.2.3.4:20622 -> x.y.z.a:12598 ...i7BX.a.'.x:@.......x......... # U x.y.z.a:12598 -> 1.2.3.4:20622 .......... Xx.........xS@.......

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Functionality for missed calls
by Miha 19 Mar '13

19 Mar '13

Hi, I need one information regarding missed calls. For exp A is talking with B. C is calling B and gets status busy. In this case it possible that B will see on his phone client C as missed calls. Why this future because users do not want second line on there phones as coustumers who are calling tham are saying that thy are not answering the phones:) so, is it possible to get missed call on phone if you are busy? tnx! miha

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registration problem 403
by Miha 19 Mar '13

19 Mar '13

Hi, all of our user agents are registereing properly. Can some help me figure out why this one is beeing rejeted? my sip trace. http://pastebin.freeswitch.org/20682 Thanks! Frame 7: 645 bytes on wire (5160 bits), 645 bytes captured (5160 bits) Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: Supermic_64:38:ff (00:30:48:64:38:ff) Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), Dst: freeswtich_ip (freeswtich_ip) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 Method: REGISTER Request-URI: sip:enterprise.freeswitch.org:5060 [Resent Packet: False] Message Header Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 Transport: UDP Sent-by Address: user_agent_ip Branch: z9hG4bK007ed7b3fda035653 Max-Forwards: 70 From: 081603006.enterprise <sip:081603006@enterprise.freeswitch.org:5060>;tag=a0ac7ed0e0 SIP Display info: 081603006.enterprise SIP from address: sip:081603006@enterprise.freeswitch.org:5060 SIP from address User Part: 081603006 SIP from address Host Part: enterprise.freeswitch.org SIP from address Host Port: 5060 SIP tag: a0ac7ed0e0 To: 081603006.enterprise <sip:081603006@enterprise.freeswitch.org:5060> SIP Display info: 081603006.enterprise SIP to address: sip:081603006@enterprise.freeswitch.org:5060 SIP to address User Part: 081603006 SIP to address Host Part: enterprise.freeswitch.org SIP to address Host Port: 5060 Call-ID: 9b78f5c9e810d200 CSeq: 1385547784 REGISTER Sequence Number: 1385547784 Method: REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO Contact: 081603006.enterprise <sip:081603006@user_agent_ip:5060;transport=udp>;expires=3600 SIP Display info: 081603006.enterprise Contact-URI: sip:081603006@user_agent_ip:5060;transport=udp Contactt-URI User Part: 081603006 Contact-URI Host Part: user_agent_ip Contact-URI Host Port: 5060 Contact parameter: transport=udp> Contact parameter: expires=3600 Supported: path User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F Content-Length: 0 No. Time Source Destination Protocol Length Info 8 153.860672 freeswtich_ip user_agent_ip SIP 751 Status: 401 Unauthorized (0 bindings) Frame 8: 751 bytes on wire (6008 bits), 751 bytes captured (6008 bits) Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: Cisco_76:c9:31 (00:16:c8:76:c9:31) Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), Dst: user_agent_ip (user_agent_ip) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Status-Code: 401 [Resent Packet: False] [Request Frame: 7] [Response Time (ms): 1] Message Header Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK007ed7b3fda035653 Transport: UDP Sent-by Address: user_agent_ip Branch: z9hG4bK007ed7b3fda035653 From: 081603006.enterprise <sip:081603006@enterprise.freeswitch.org:5060>;tag=a0ac7ed0e0 SIP Display info: 081603006.enterprise SIP from address: sip:081603006@enterprise.freeswitch.org:5060 SIP from address User Part: 081603006 SIP from address Host Part: enterprise.freeswitch.org SIP from address Host Port: 5060 SIP tag: a0ac7ed0e0 To: 081603006.enterprise <sip:081603006@enterprise.freeswitch.org:5060>;tag=2cr6m0BZBUUKg SIP Display info: 081603006.enterprise SIP to address: sip:081603006@enterprise.freeswitch.org:5060 SIP to address User Part: 081603006 SIP to address Host Part: enterprise.freeswitch.org SIP to address Host Port: 5060 SIP tag: 2cr6m0BZBUUKg Call-ID: 9b78f5c9e810d200 CSeq: 1385547784 REGISTER Sequence Number: 1385547784 Method: REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="enterprise.freeswitch.org", nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4", algorithm=MD5, qop="auth" Authentication Scheme: Digest realm="enterprise.freeswitch.org" nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" algorithm=MD5 qop="auth" Content-Length: 0 No. Time Source Destination Protocol Length Info 9 153.875842 user_agent_ip freeswtich_ip SIP 917 Request: REGISTER sip:enterprise.freeswitch.org:5060 Frame 9: 917 bytes on wire (7336 bits), 917 bytes captured (7336 bits) Ethernet II, Src: Cisco_76:c9:31 (00:16:c8:76:c9:31), Dst: Supermic_64:38:ff (00:30:48:64:38:ff) Internet Protocol Version 4, Src: user_agent_ip (user_agent_ip), Dst: freeswtich_ip (freeswtich_ip) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:enterprise.freeswitch.org:5060 SIP/2.0 Method: REGISTER Request-URI: sip:enterprise.freeswitch.org:5060 [Resent Packet: False] Message Header Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 Transport: UDP Sent-by Address: user_agent_ip Branch: z9hG4bK1cc3b9f75e01d9b85 Max-Forwards: 70 From: 081603006.enterprise <sip:081603006@enterprise.freeswitch.org:5060>;tag=a0ac7ed0e0 SIP Display info: 081603006.enterprise SIP from address: sip:081603006@enterprise.freeswitch.org:5060 SIP from address User Part: 081603006 SIP from address Host Part: enterprise.freeswitch.org SIP from address Host Port: 5060 SIP tag: a0ac7ed0e0 To: 081603006.enterprise <sip:081603006@enterprise.freeswitch.org:5060> SIP Display info: 081603006.enterprise SIP to address: sip:081603006@enterprise.freeswitch.org:5060 SIP to address User Part: 081603006 SIP to address Host Part: enterprise.freeswitch.org SIP to address Host Port: 5060 Call-ID: 9b78f5c9e810d200 CSeq: 1385547785 REGISTER Sequence Number: 1385547785 Method: REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO [truncated] Authorization: Digest username="081603006.enterprise",realm="enterprise.freeswitch.org",nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4",uri="sip:enterprise.freeswitch.org:5060",response="c9fbce339bb5aa1137ea34bf190619ac",algorithm Authentication Scheme: Digest username="081603006.enterprise" realm="enterprise.freeswitch.org" nonce="140088b8-c23a-461e-abb0-2ceef4bb16f4" uri="sip:enterprise.freeswitch.org:5060" response="c9fbce339bb5aa1137ea34bf190619ac" algorithm=MD5 qop=auth cnonce="41fb0f25" nc=00000001 Contact: 081603006.enterprise <sip:081603006@user_agent_ip:5060;transport=udp>;expires=3600 SIP Display info: 081603006.enterprise Contact-URI: sip:081603006@user_agent_ip:5060;transport=udp Contactt-URI User Part: 081603006 Contact-URI Host Part: user_agent_ip Contact-URI Host Port: 5060 Contact parameter: transport=udp> Contact parameter: expires=3600 Supported: path User-Agent: ARRIS-TM902S release v.7.10.145.SIP SN/001DCE73F07F Content-Length: 0 No. Time Source Destination Protocol Length Info 10 153.876474 freeswtich_ip user_agent_ip SIP 615 Status: 403 Forbidden (0 bindings) Frame 10: 615 bytes on wire (4920 bits), 615 bytes captured (4920 bits) Ethernet II, Src: Supermic_64:38:ff (00:30:48:64:38:ff), Dst: Cisco_76:c9:31 (00:16:c8:76:c9:31) Internet Protocol Version 4, Src: freeswtich_ip (freeswtich_ip), Dst: user_agent_ip (user_agent_ip) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 403 Forbidden Status-Code: 403 [Resent Packet: False] [Request Frame: 9] [Response Time (ms): 0] Message Header Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK1cc3b9f75e01d9b85 Transport: UDP Sent-by Address: user_agent_ip Branch: z9hG4bK1cc3b9f75e01d9b85 From: 081603006.enterprise <sip:081603006@enterprise.freeswitch.org:5060>;tag=a0ac7ed0e0 SIP Display info: 081603006.enterprise SIP from address: sip:081603006@enterprise.freeswitch.org:5060 SIP from address User Part: 081603006 SIP from address Host Part: enterprise.freeswitch.org SIP from address Host Port: 5060 SIP tag: a0ac7ed0e0 To: 081603006.enterprise <sip:081603006@enterprise.freeswitch.org:5060>;tag=3NHZpUv283H6B SIP Display info: 081603006.enterprise SIP to address: sip:081603006@enterprise.freeswitch.org:5060 SIP to address User Part: 081603006 SIP to address Host Part: enterprise.freeswitch.org SIP to address Host Port: 5060 SIP tag: 3NHZpUv283H6B Call-ID: 9b78f5c9e810d200 CSeq: 1385547785 REGISTER Sequence Number: 1385547785 Method: REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20121004T033301Z~94664868a8 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0

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Table lookups in a dialplan - is using mod_blacklist appropriate or too much of a hack?
by Nathan Neulinger 18 Mar '13

18 Mar '13

I was thinking of using mod_blacklist as a way to define a table of short extension numbers to be able to "look up" an entry in a dialplan condition. i.e. defining a blacklist "local_ext_but_not_freeswitch" - or for distinguishing easily between sip and skinny phones (is_sip_ext, is_skinny_ext). Does this seem like a useful use of the (somewhat misnamed) mod_blacklist, or a bad approach? Reason for this is that I'll have approximately 1600 extensions, discontiguous - some of which will be on freeswitch, others which won't. For the ones that are on freeswitch, some of them will be on skinny phones, and others will be on SIP phones. I was leaning toward something like: <extension name="Local Ext Check"> <condition field="${blacklist(check is_local_ext $1)}" expression="^true$"> <condition field="destination_number" expression="^(\d\d\d\d)$"> </condition> </condition> </extension> The alternative was to build a new module very similar to mod_blacklist - i.e. 'mod_lookup', with the additional capability of having a value lookup and not just existence check. I realize I could rig something up with mod_hash, but I like the 'load straight from a file' capability. Thoughts? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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E1 card suggestion
by Tomasz Szuster 18 Mar '13

18 Mar '13

Hi Guys, I'm wondering if you can advice a good card for E1 connections. Which is working very well under freeswitch and installation and configuration will not make my head to explode ;) -- Regards. Tom

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