[asterisk-users] Diagnosing call problem

Mitch Claborn mitch_ml at claborn.net
Wed Mar 20 12:55:00 CDT 2013


That change did not fix the problem.  Below is another trace from a 
failed call this morning.  172.16.0.71 is the client, 172.16.0.245 is 
the Asterisk server.  All the RTP packets after the SIP are from server 
to client.

Any further ideas are appreciated.  (If I don't get this fixed this 
week, I won't get to go home on Friday!)

-----------------------------------------------

No.     Time            Source                Destination 
Protocol Length Info
    4528 12:14:07.219165 172.16.0.245          172.16.0.71 
SIP/SDP  910    Request: INVITE sip:KWakmn at 172.16.0.71:5060, with 
session description

Frame 4528: 910 bytes on wire (7280 bits), 910 bytes captured (7280 bits)
Ethernet II, Src: 90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35), Dst: 
Dell_e7:fc:b0 (00:25:64:e7:fc:b0)
Internet Protocol Version 4, Src: 172.16.0.245 (172.16.0.245), Dst: 
172.16.0.71 (172.16.0.71)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
     Request-Line: INVITE sip:KWakmn at 172.16.0.71:5060 SIP/2.0
     Message Header
         Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK7e5fc96a
         Max-Forwards: 70
         From: <sip:4062345243 at 172.16.0.245>;tag=as5a63ac9a
         To: <sip:KWakmn at 172.16.0.71:5060>
         Contact: <sip:4062345243 at 172.16.0.245:5060>
         Call-ID: 50b7a1e27bbb9f6043dfccff16d7be88 at 172.16.0.245:5060
         CSeq: 102 INVITE
         User-Agent: Asterisk PBX 11.1.0
         Date: Wed, 20 Mar 2013 17:14:07 GMT
         Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY, INFO, PUBLISH
         Supported: replaces, timer
         X-mm-call: http://www.mcmurrayhatchery.com
         Content-Type: application/sdp
         Content-Length: 257
     Message Body
         Session Description Protocol
             Session Description Protocol Version (v): 0
             Owner/Creator, Session Id (o): root 582679053 582679053 IN 
IP4 172.16.0.245
             Session Name (s): Asterisk PBX 11.1.0
             Connection Information (c): IN IP4 172.16.0.245
             Time Description, active time (t): 0 0
             Media Description, name and address (m): audio 28340 
RTP/AVP 0 8 101
             Media Attribute (a): rtpmap:0 PCMU/8000
             Media Attribute (a): rtpmap:8 PCMA/8000
             Media Attribute (a): rtpmap:101 telephone-event/8000
             Media Attribute (a): fmtp:101 0-16
             Media Attribute (a): ptime:20
             Media Attribute (a): sendrecv

------------------------------------------

No.     Time            Source                Destination 
Protocol Length Info
       1 12:14:07.251118 172.16.0.71           172.16.0.245          SIP 
      542    Status: 180 Ringing

Frame 1: 542 bytes on wire (4336 bits), 542 bytes captured (4336 bits)
Ethernet II, Src: Dell_e7:fc:b0 (00:25:64:e7:fc:b0), Dst: 
90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35)
Internet Protocol Version 4, Src: 172.16.0.71 (172.16.0.71), Dst: 
172.16.0.245 (172.16.0.245)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
     Status-Line: SIP/2.0 180 Ringing
     Message Header
         Via: SIP/2.0/UDP 
172.16.0.245:5060;received=172.16.0.245;branch=z9hG4bK7e5fc96a
         Call-ID: 50b7a1e27bbb9f6043dfccff16d7be88 at 172.16.0.245:5060
         From: <sip:4062345243 at 172.16.0.245>;tag=as5a63ac9a
         To: 
<sip:KWakmn at 172.16.0.71>;tag=923b9add-5ef2-4f6f-a3f5-4627109e079c
         CSeq: 102 INVITE
         Contact: <sip:KWakmn at 172.16.0.71:5060>
         Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, 
CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE, INVITE, ACK, BYE, CANCEL
         Content-Length:  0

------------------------------------------

No.     Time            Source                Destination 
Protocol Length Info
       5 12:14:18.055112 172.16.0.71           172.16.0.245 
SIP/SDP  834    Status: 200 OK, with session description

Frame 5: 834 bytes on wire (6672 bits), 834 bytes captured (6672 bits)
Ethernet II, Src: Dell_e7:fc:b0 (00:25:64:e7:fc:b0), Dst: 
90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35)
Internet Protocol Version 4, Src: 172.16.0.71 (172.16.0.71), Dst: 
172.16.0.245 (172.16.0.245)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
     Status-Line: SIP/2.0 200 OK
     Message Header
         Via: SIP/2.0/UDP 
172.16.0.245:5060;received=172.16.0.245;branch=z9hG4bK7e5fc96a
         Call-ID: 50b7a1e27bbb9f6043dfccff16d7be88 at 172.16.0.245:5060
         From: <sip:4062345243 at 172.16.0.245>;tag=as5a63ac9a
         To: 
<sip:KWakmn at 172.16.0.71>;tag=923b9add-5ef2-4f6f-a3f5-4627109e079c
         CSeq: 102 INVITE
         Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, 
CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE, INVITE, ACK, BYE, CANCEL
         Contact: <sip:KWakmn at 172.16.0.71:5060>
         Supported: replaces, 100rel
         Content-Type: application/sdp
         Content-Length:   234
     Message Body
         Session Description Protocol
             Session Description Protocol Version (v): 0
             Owner/Creator, Session Id (o): asset071 3572788447 1 IN IP4 
172.16.0.71
             Session Name (s): sflphone
             Connection Information (c): IN IP4 172.16.0.71
             Time Description, active time (t): 0 0
             Media Description, name and address (m): audio 45208 RTP/AVP 0
             Media Attribute (a): rtpmap:0 PCMU/8000
             Media Attribute (a): sendrecv
             Media Attribute (a): rtpmap:101 telephone-event/8000
             Media Attribute (a): fmtp:101 0-15
             Media Attribute (a): rtcp:45209 IN IP4 172.16.0.71

----------------------------------

No.     Time            Source                Destination 
Protocol Length Info
       6 12:14:18.056116 172.16.0.245          172.16.0.71           SIP 
      463    Request: ACK sip:KWakmn at 172.16.0.71:5060

Frame 6: 463 bytes on wire (3704 bits), 463 bytes captured (3704 bits)
Ethernet II, Src: 90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35), Dst: 
Dell_e7:fc:b0 (00:25:64:e7:fc:b0)
Internet Protocol Version 4, Src: 172.16.0.245 (172.16.0.245), Dst: 
172.16.0.71 (172.16.0.71)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
     Request-Line: ACK sip:KWakmn at 172.16.0.71:5060 SIP/2.0
     Message Header
         Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK5a4eafc5
         Max-Forwards: 70
         From: <sip:4062345243 at 172.16.0.245>;tag=as5a63ac9a
         To: 
<sip:KWakmn at 172.16.0.71:5060>;tag=923b9add-5ef2-4f6f-a3f5-4627109e079c
         Contact: <sip:4062345243 at 172.16.0.245:5060>
         Call-ID: 50b7a1e27bbb9f6043dfccff16d7be88 at 172.16.0.245:5060
         CSeq: 102 ACK
         User-Agent: Asterisk PBX 11.1.0
         Content-Length: 0

----------


Mitch

On 03/19/2013 07:18 PM, Mitch Claborn wrote:
> Good point. I changed to 10000 - 40000.
>
>
> Mitch
>
> On 03/19/2013 06:17 PM, Asghar Mohammad wrote:
>> i had this problem with a gateway witch was configured from 1000 to 3000
>> and some time he was using ports above 2000 and result was one way voice
>> rtp port range is where asterisk expect audio, you should not use ports
>> below 10000 because they are in use of other services like 5060 for sip.
>>
>> On Tue, Mar 19, 2013 at 11:57 PM, Mitch Claborn <mitch_ml at claborn.net
>> <mailto:mitch_ml at claborn.net>> wrote:
>>
>>     This was the client sending from port 39409 to server port 13429,
>>     which is in the range.  From what I read, the rtpstart and rtpend
>>     define the range that is available for use on the server, so I'm not
>>     sure this will apply.
>>
>>     But, I've set my range to 5000 - 40000.  I'll find out tomorrow if
>>     it makes any difference.
>>
>>     Where is a good place to find documentation on the various fields in
>>     the INVITE SIP message and the response? I'd like to dig into them
>>     and try to understand them more completely.
>>
>>
>>     Mitch
>>
>>
>>     On 03/19/2013 05:02 PM, Asghar Mohammad wrote:
>>
>>         hi,
>>
>>         "User Datagram Protocol, Src Port: 39409 (39409), Dst Port:
>>         13429 (13429)"
>>
>>         copy from asterisk 11 rtp.conf
>>         rtpstart=10000
>>         rtpend=20000
>>
>>         have you changed port range? if no then
>>         your client sending rtp to a port higher then configured in
>> rtp port
>>         range and asterisk ignore that port.
>>         try to change rtpend=30000 or if there is option in
>>         softphone restrict it to use same range as in rtp.conf.
>>
>>         let me know if this solve you problem.
>>
>>         On Tue, Mar 19, 2013 at 10:54 PM, Asghar Mohammad
>>         <asghar144 at gmail.com <mailto:asghar144 at gmail.com>
>>         <mailto:asghar144 at gmail.com <mailto:asghar144 at gmail.com>>> wrote:
>>
>>              hi,
>>
>>              "User Datagram Protocol, Src Port: 39409 (39409), Dst Port:
>>         13429
>>              (13429)"
>>
>>              copy from asterisk 11 rtp.conf
>>              rtpstart=10000
>>              rtpend=20000
>>
>>              have you changed port range? if no then
>>              your client sending rtp to a port higher then configured in
>>         rtp port
>>              range and asterisk ignore that port.
>>              try to change rtpend=30000 or if there is option in
>>              softphone restrict it to use same range as in rtp.conf.
>>
>>              let me know if this solve you problem.
>>
>>
>>              On Tue, Mar 19, 2013 at 10:22 PM, Mitch Claborn
>>              <mitch_ml at claborn.net <mailto:mitch_ml at claborn.net>
>>         <mailto:mitch_ml at claborn.net <mailto:mitch_ml at claborn.net>>>
>> wrote:
>>
>>                  We have Ubuntu 12.04 clients, using either SFLPhone or
>>         Bria 3.
>>                  There is no NAT involved in the network at all (it is
>>         disabled
>>                  in sip.conf).
>>
>>                  Here are the SIP messages capture via wireshark on the
>>         client
>>                  during one problem call.  Following these SIP
>> messages, the
>>                  wireshark trace shows only RTP packets from server
>>                  (172.16.0.245) to client (172.16.0.71) except for an
>>         occasional
>>                  RTCP packet from client to server (sample below).
>>
>>                  Any help is appreciated. The uses are really beating me
>>         up to
>>                  get this fixed.
>>
>>                  --------------------
>>
>>                  INVITE sip:KWakmn at 172.16.0.71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>
>>                  <http://sip:KWakmn@172.16.0.__71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>> SIP/2.0
>>                  Via: SIP/2.0/UDP
>>         172.16.0.245:5060;branch=____z9hG4bK19e2246d
>>
>>                  Max-Forwards: 70
>>                  From: <sip:2392230612 at 172.16.0.245
>>         <mailto:sip%3A2392230612 at 172.16.0.245>
>>                  <mailto:sip%3A2392230612 at 172.__16.0.245
>>         <mailto:sip%253A2392230612 at 172.16.0.245>>>;__tag=as4b489afc
>>                  To: <sip:KWakmn at 172.16.0.71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>
>>                  <http://sip:KWakmn@172.16.0.__71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>>>
>>                  Contact: <sip:2392230612
>>         <tel:2392230612>@172.16.0.245:__5060
>>                  <http://sip:2392230612@172.16.__0.245:5060
>>         <http://sip:2392230612@172.16.0.245:5060>>>
>>                  Call-ID:
>>         52106f231b41169c7eabd3b43d0fc6____e8 at 172.16.0.245:5060
>>         <http://52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060>
>>
>>         <http://__52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060
>>         <http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060>>
>>
>>                  CSeq: 102 INVITE
>>                  User-Agent: Asterisk PBX 11.1.0
>>                  Date: Tue, 19 Mar 2013 20:47:26 GMT
>>                  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>> SUBSCRIBE,
>>                  NOTIFY, INFO, PUBLISH
>>                  Supported: replaces, timer
>>                  X-mm-call: http://www.mcmurrayhatchery.____com
>>
>>                  <http://www.mcmurrayhatchery.__com
>>         <http://www.mcmurrayhatchery.com>>
>>                  Content-Type: application/sdp
>>                  Content-Length: 257
>>
>>                  v=0
>>                  o=root 682517197 682517197 IN IP4 172.16.0.245
>>                  s=Asterisk PBX 11.1.0
>>                  c=IN IP4 172.16.0.245
>>                  t=0 0
>>                  m=audio 13428 RTP/AVP 0 8 101
>>                  a=rtpmap:0 PCMU/8000
>>                  a=rtpmap:8 PCMA/8000
>>                  a=rtpmap:101 telephone-event/8000
>>                  a=fmtp:101 0-16
>>                  a=ptime:20
>>                  a=sendrecv
>>
>>                  ------------------------------____-
>>
>>
>>                  SIP/2.0 180 Ringing
>>                  Via: SIP/2.0/UDP
>>
>>
>> 172.16.0.245:5060;received=____172.16.0.245;branch=____z9hG4bK19e2246d
>>                  Call-ID:
>>         52106f231b41169c7eabd3b43d0fc6____e8 at 172.16.0.245:5060
>>         <http://52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060>
>>
>>         <http://__52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060
>>         <http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060>>
>>                  From: <sip:2392230612 at 172.16.0.245
>>         <mailto:sip%3A2392230612 at 172.16.0.245>
>>                  <mailto:sip%3A2392230612 at 172.__16.0.245
>>         <mailto:sip%253A2392230612 at 172.16.0.245>>>;__tag=as4b489afc
>>                  To: <sip:KWakmn at 172.16.0.71
>>         <mailto:sip%3AKWakmn at 172.16.0.71>
>>                  <mailto:sip%3AKWakmn at 172.16.0.__71
>>
>> <mailto:sip%253AKWakmn at 172.16.0.71>>>;tag=__7543f39a-7ca0-434b-__8281-__e6dc2adc4aa3
>>
>>
>>                  CSeq: 102 INVITE
>>                  Contact: <sip:KWakmn at 172.16.0.71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>
>>                  <http://sip:KWakmn@172.16.0.__71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>>>
>>
>>                  Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK,
>> BYE,
>>                  CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE,
>>         INVITE, ACK,
>>                  BYE, CANCEL
>>                  Content-Length: 0
>>
>>
>> ------------------------------____-----------------------
>>
>>
>>                  SIP/2.0 200 OK
>>                  Via: SIP/2.0/UDP
>>
>>
>> 172.16.0.245:5060;received=____172.16.0.245;branch=____z9hG4bK19e2246d
>>                  Call-ID:
>>         52106f231b41169c7eabd3b43d0fc6____e8 at 172.16.0.245:5060
>>         <http://52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060>
>>
>>         <http://__52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060
>>         <http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060>>
>>                  From: <sip:2392230612 at 172.16.0.245
>>         <mailto:sip%3A2392230612 at 172.16.0.245>
>>                  <mailto:sip%3A2392230612 at 172.__16.0.245
>>         <mailto:sip%253A2392230612 at 172.16.0.245>>>;__tag=as4b489afc
>>                  To: <sip:KWakmn at 172.16.0.71
>>         <mailto:sip%3AKWakmn at 172.16.0.71>
>>                  <mailto:sip%3AKWakmn at 172.16.0.__71
>>
>> <mailto:sip%253AKWakmn at 172.16.0.71>>>;tag=__7543f39a-7ca0-434b-__8281-__e6dc2adc4aa3
>>
>>
>>                  CSeq: 102 INVITE
>>                  Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK,
>> BYE,
>>                  CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE,
>>         INVITE, ACK,
>>                  BYE, CANCEL
>>                  Contact: <sip:KWakmn at 172.16.0.71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>
>>                  <http://sip:KWakmn@172.16.0.__71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>>>
>>
>>                  Supported: replaces, 100rel
>>                  Content-Type: application/sdp
>>                  Content-Length: 234
>>
>>                  v=0
>>                  o=asset071 3572714846 1 IN IP4 172.16.0.71
>>                  s=sflphone
>>                  c=IN IP4 172.16.0.71
>>                  t=0 0
>>                  m=audio 39408 RTP/AVP 0
>>                  a=rtpmap:0 PCMU/8000
>>                  a=sendrecv
>>                  a=rtpmap:101 telephone-event/8000
>>                  a=fmtp:101 0-15
>>                  a=rtcp:39409 IN IP4 172.16.0.71
>>
>>                  ------------------------------____-----------------
>>
>>                  ACK sip:KWakmn at 172.16.0.71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>
>>                  <http://sip:KWakmn@172.16.0.__71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>> SIP/2.0
>>                  Via: SIP/2.0/UDP
>>         172.16.0.245:5060;branch=____z9hG4bK289d6da2
>>
>>                  Max-Forwards: 70
>>                  From: <sip:2392230612 at 172.16.0.245
>>         <mailto:sip%3A2392230612 at 172.16.0.245>
>>                  <mailto:sip%3A2392230612 at 172.__16.0.245
>>         <mailto:sip%253A2392230612 at 172.16.0.245>>>;__tag=as4b489afc
>>                  To: <sip:KWakmn at 172.16.0.71:5060
>>         <http://sip:KWakmn@172.16.0.71:5060>
>>                  <http://sip:KWakmn@172.16.0.__71:5060
>>
>> <http://sip:KWakmn@172.16.0.71:5060>>>;__tag=7543f39a-7ca0-__434b-8281-__e6dc2adc4aa3
>>
>>                  Contact: <sip:2392230612
>>         <tel:2392230612>@172.16.0.245:__5060
>>                  <http://sip:2392230612@172.16.__0.245:5060
>>         <http://sip:2392230612@172.16.0.245:5060>>>
>>                  Call-ID:
>>         52106f231b41169c7eabd3b43d0fc6____e8 at 172.16.0.245:5060
>>         <http://52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060>
>>
>>         <http://__52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060
>>         <http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060>>
>>
>>                  CSeq: 102 ACK
>>                  User-Agent: Asterisk PBX 11.1.0
>>                  Content-Length: 0
>>
>>
>>
>> ------------------------------____----------------------------__--
>>
>>
>>                  SAMPLE RTCP packet from client to server
>>
>>                  No.     Time            Source
>> Destination
>>                  Protocol Length Info
>>                       240 15:47:39.965483 172.16.0.71
>>         172.16.0.245 RTCP
>>                       102    Receiver Report   Source description
>>
>>                  Frame 240: 102 bytes on wire (816 bits), 102 bytes
>>         captured (816
>>                  bits)
>>                  Ethernet II, Src: Dell_e7:fc:b0 (00:25:64:e7:fc:b0),
>> Dst:
>>                  90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35)
>>                  Internet Protocol Version 4, Src: 172.16.0.71
>>         (172.16.0.71),
>>                  Dst: 172.16.0.245 (172.16.0.245)
>>                  User Datagram Protocol, Src Port: 39409 (39409), Dst
>>         Port: 13429
>>                  (13429)
>>                  Real-time Transport Control Protocol (Receiver Report)
>>                       [Stream setup by SDP (frame 36)]
>>                           [Setup frame: 36]
>>                           [Setup Method: SDP]
>>                       10.. .... = Version: RFC 1889 Version (2)
>>                       ..0. .... = Padding: False
>>                       ...0 0001 = Reception report count: 1
>>                       Packet type: Receiver Report (201)
>>                       Length: 7 (32 bytes)
>>                       Sender SSRC: 0x841ef2ea (2216620778)
>>                       Source 1
>>                           Identifier: 0x28bcc3a6 (683459494)
>>                           SSRC contents
>>                               Fraction lost: 254 / 256
>>                               Cumulative number of packets lost: 37134
>>                           Extended highest sequence number received:
>> 37331
>>                               Sequence number cycles count: 0
>>                               Highest sequence number received: 37331
>>                           Interarrival jitter: 160008128
>>                           Last SR timestamp: 0 (0x00000000)
>>                           Delay since last SR timestamp: 0 (0
>> milliseconds)
>>                  Real-time Transport Control Protocol (Source
>> description)
>>                       [Stream setup by SDP (frame 36)]
>>                           [Setup frame: 36]
>>                           [Setup Method: SDP]
>>                       10.. .... = Version: RFC 1889 Version (2)
>>                       ..0. .... = Padding: False
>>                       ...0 0001 = Source count: 1
>>                       Packet type: Source description (202)
>>                       Length: 6 (28 bytes)
>>                       Chunk 1, SSRC/CSRC 0x841EF2EA
>>                           Identifier: 0x841ef2ea (2216620778)
>>                           SDES items
>>                               Type: CNAME (user and domain) (1)
>>                               Length: 17
>>                               Text: kristin at localhost
>>                               Type: END (0)
>>                  [RTCP frame length check: OK - 60 bytes]
>>
>>
>>
>>
>>
>>                  Mitch
>>
>>
>>
>>         --
>>
>> _________________________________________________________________________
>>         -- Bandwidth and Colocation Provided by
>>         http://www.api-digital.com --
>>         New to Asterisk? Join us for a live introductory webinar every
>>         Thurs:
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>>     --
>>
>> _________________________________________________________________________
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>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
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