[cisco-voip] 78xx phones in SRST mode - Stops working after about 20-30 minutes, other phone models continue to work
Brian Meade
bmeade90 at vt.edu
Wed Dec 21 13:13:17 EST 2016
Looks like maybe this bug
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCut30586
It's opened as a feature enhancement request.
Do the console logs and a packet capture from the phone side show it
receiving the invite immediately. Do you see it sending trying to send
Register messages to CUCM that it's waiting on in that time?
On Wed, Dec 21, 2016 at 12:57 PM, Brian V <bvanbens at gmail.com> wrote:
> I've having a strange issue with 78xx phones while in SRST mode.
>
> I've isolated it on the bench and I'm able to reproduce it.
>
> I have a TAC case open but wondering if anyone has run into anything
> similar?
>
> have 3 phones for testing.
>
> 7821 SIP phone load: 11-0-1-11
>
> 7965 SCCP phone (SCCP45.9-4-2SR1)
>
> 8961 SIP phone - older one but with newer code (sip8961.9.4.2 SR2)
>
> Router/IOS: 4321 - Denali 16.03.02
>
> CUCM version 10.5.2 SU2
>
>
>
> Typical remote CUCM site configured for SRST. Mix of 7821, 8961, and
> 7965 phones.
>
> Router is configured for both SCCP and SIP SRST.
>
> When WAN connection is lost, phones register with router correctly after a
> few minutes and function properly. Can make calls between them all, and
> calls to/from PSTN
>
> After a period of time (about 20-30 min) the 78xx phones seem to stop
> working. Can't make calls TO them and can't make calls FROM them. During
> this time calls work between the 8961 phones and the 7965 phones and
> to/from the PSTN.
>
>
>
>
>
> During this time where the 7821 phones don’t work, the output of the *show
> voice register* commands shows the 7821 phone is registered with the
> auto-created dial-peers still present.
>
>
>
> I enabled SIP message debugging, this is where it gets interesting.
>
> Test is to place a call TO the 7821 phone. From SCCP ext. 2014 to
> SIP(7821 phones) ext. 2004
>
> I placed a call from the SCCP phone to eliminate one side of the SIP
> messages but the result is the same regardless if it place the call from an
> 8961 or 79xx phone.
>
>
>
>
>
> I see the immediately router send out 3 invites to the 7821 phone
>
> During this time, the phone doesn’t show any activity.
>
> Then about 1 full minute later we see the 7821 phone send back three
> 100-Trying messages and three 180-ringing messages
>
> During this time the 7821 phone is now ringing a full minute after I
> already hung up the calling phone
>
> I pick up the handset on the 7821 phone, there is not an actual call there.
>
> I hang it up
>
> Then about 2 minutes later the 7821 phone sends back 200-OK messages with
> SDP. There was no activity on the phone during this time.
>
>
>
> It almost "seems" like the 7821 phone is having high CPU usage and its
> super slow in responding to SIP messages.
>
> This state take a while to develop. Immediately after the phones switch
> into SRST mode they work fine , then after a period of time, just the 78xx
> phones exhibit this behavior.
>
>
>
> See debugs and config below
>
> SRST IP: 172.25.4.200
>
> Target phone IP: 172.25.4.26
>
>
>
> voice service voip
>
> ip address trusted list
>
> ipv4 10.0.0.0 255.0.0.0
>
> ipv4 192.168.0.0 255.255.0.0
>
> ipv4 10.10.111.10 255.255.255.255
>
> ipv4 172.16.0.0 255.240.0.0
>
> media statistics
>
> allow-connections sip to sip
>
> fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
>
> call-quality
>
> sip
>
> bind control source-interface GigabitEthernet0/0/0
>
> bind media source-interface GigabitEthernet0/0/0
>
> min-se 600
>
> registrar server expires max 1800 min 900
>
> no call service stop
>
>
>
> voice register global
>
> default mode
>
> no allow-hash-in-dn
>
> timeouts interdigit 5
>
> system message SIP SRST Mode
>
> max-dn 25
>
> max-pool 10
>
> timezone 8
>
> !
>
> voice register pool 1
>
> id network 172.25.4.0 mask 255.255.255.0
>
> dtmf-relay rtp-nte sip-kpml
>
> codec g711ulaw
>
> no vad
>
> !
>
> voice hunt-group 1 parallel
>
> list 2004,3003,2001,2014
>
> pilot 2424
>
>
>
> interface GigabitEthernet0/0/0
>
> ip address 172.25.4.200 255.255.255.0
>
> media-type rj45
>
> negotiation auto
>
>
>
> call-manager-fallback
>
> secondary-dialtone 9
>
> max-conferences 8 gain -6
>
> transfer-system full-consult
>
> timeouts interdigit 5
>
> ip source-address 172.25.4.200 port 2000
>
> max-ephones 10
>
> max-dn 25 octo-line
>
> system message primary SCCP SRST MODE
>
> time-zone 8
>
>
>
> TEST-CUBE#
>
> TEST-CUBE# sh debug
>
> IOSXE Conditional Debug Configs:
>
>
>
> Conditional Debug Global State: Stop
>
>
>
>
>
> IOSXE Packet Tracing Configs:
>
>
>
> Packet Infra debugs:
>
>
>
> Ip Address Port
>
> ------------------------------------------------------|----------
>
> CCSIP SPI: SIP Call Message tracing is enabled (filter is OFF)
>
> CCSIP SPI: SIP Out-of-Dialog tracing is enabled (filter is OFF)
>
>
>
> voice-register:
>
> voice-register error debugging is on
>
> voice-register events debugging is on
>
>
>
> TEST-CUBE#
>
>
>
>
>
> !! ## this event is about 20-30 min after the phone went into SRST mode
> and registered with the router
>
> !! ## prior to this time, it functioned normally.
>
> !!#### make call from SCCP phone extension 2001 to SIP phone extension
> 2004
>
> !!### Sends 3 invites is rapid succession. Note time stamps
>
>
>
> 030590: .Dec 20 *17:38:16.801*: //3480/365A43AF9107/SIP/Msg/
> ccsipDisplayMsg:
>
> Sent:
>
> INVITE sip:2004 at 172.25.4.26:5060 SIP/2.0
>
> Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
>
> Remote-Party-ID: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200
> >;party=calling;screen=no;privacy=off
>
> From: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200>;tag=
> 5E6EC01-2547
>
> To: <sip:2004 at 172.25.4.26>
>
> Date: Tue, 20 Dec 2016 23:38:16 GMT
>
> Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
>
> Supported: 100rel,timer,resource-priority,replaces
>
> Min-SE: 600
>
> Cisco-Guid: 0911885231-3326349798-2433210144-2638814607
>
> User-Agent: Cisco-SIPGateway/IOS-16.3.2
>
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
> NOTIFY, INFO, REGISTER
>
> CSeq: 101 INVITE
>
> Max-Forwards: 70
>
> Timestamp: 1482277096
>
> Contact: <sip:2001 at 172.25.4.200:5060>
>
> Expires: 180
>
> Allow-Events: kpml, telephone-event
>
> Session-ID: e3d1d5b9cf5d5821bc81fbad4de46749;remote=
> 00000000000000000000000000000000
>
> Content-Type: application/sdp
>
> Content-Disposition: session;handling=required
>
> Content-Length: 246
>
>
>
> v=0
>
> o=CiscoSystemsSIP-GW-UserAgent 4428 5277 IN IP4 172.25.4.200
>
> s=SIP Call
>
> c=IN IP4 172.25.4.200
>
> t=0 0
>
> m=audio 8508 RTP/AVP 0 101
>
> c=IN IP4 172.25.4.200
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=ptime:20
>
>
>
> 030591: .Dec 20 17:38:17.051: //3480/365A43AF9107/SIP/Msg/ccsipDisplayMsg:
>
> Sent:
>
> INVITE sip:2004 at 172.25.4.26:5060 SIP/2.0
>
> Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
>
> Remote-Party-ID: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200
> >;party=calling;screen=no;privacy=off
>
> From: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200>;tag=
> 5E6EC01-2547
>
> To: <sip:2004 at 172.25.4.26>
>
> Date: Tue, 20 Dec 2016 23:38:17 GMT
>
> Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
>
> Supported: 100rel,timer,resource-priority,replaces
>
> Min-SE: 600
>
> Cisco-Guid: 0911885231-3326349798-2433210144-2638814607
>
> User-Agent: Cisco-SIPGateway/IOS-16.3.2
>
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
> NOTIFY, INFO, REGISTER
>
> CSeq: 101 INVITE
>
> Max-Forwards: 70
>
> Timestamp: 1482277097
>
> Contact: <sip:2001 at 172.25.4.200:5060>
>
> Expires: 180
>
> Allow-Events: kpml, telephone-event
>
> Session-ID: e3d1d5b9cf5d5821bc81fbad4de46749;remote=
> 00000000000000000000000000000000
>
> Content-Type: application/sdp
>
> Content-Disposition: session;handling=required
>
> Content-Length: 246
>
>
>
> v=0
>
> o=CiscoSystemsSIP-GW-UserAgent 4428 5277 IN IP4 172.25.4.200
>
> s=SIP Call
>
> c=IN IP4 172.25.4.200
>
> t=0 0
>
> m=audio 8508 RTP/AVP 0 101
>
> c=IN IP4 172.25.4.200
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=ptime:20
>
>
>
> 030592: .Dec 20 17:38:17.550: //3480/365A43AF9107/SIP/Msg/ccsipDisplayMsg:
>
> Sent:
>
> INVITE sip:2004 at 172.25.4.26:5060 SIP/2.0
>
> Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
>
> Remote-Party-ID: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200
> >;party=calling;screen=no;privacy=off
>
> From: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200>;tag=
> 5E6EC01-2547
>
> To: <sip:2004 at 172.25.4.26>
>
> Date: Tue, 20 Dec 2016 23:38:17 GMT
>
> Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
>
> Supported: 100rel,timer,resource-priority,replaces
>
> Min-SE: 600
>
> Cisco-Guid: 0911885231-3326349798-2433210144-2638814607
>
> User-Agent: Cisco-SIPGateway/IOS-16.3.2
>
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
> NOTIFY, INFO, REGISTER
>
> CSeq: 101 INVITE
>
> Max-Forwards: 70
>
> Timestamp: 1482277097
>
> Contact: <sip:2001 at 172.25.4.200:5060>
>
> Expires: 180
>
> Allow-Events: kpml, telephone-event
>
> Session-ID: e3d1d5b9cf5d5821bc81fbad4de46749;remote=
> 00000000000000000000000000000000
>
> Content-Type: application/sdp
>
> Content-Disposition: session;handling=required
>
> Content-Length: 246
>
>
>
> v=0
>
> o=CiscoSystemsSIP-GW-UserAgent 4428 5277 IN IP4 172.25.4.200
>
> s=SIP Call
>
> c=IN IP4 172.25.4.200
>
> t=0 0
>
> m=audio 8508 RTP/AVP 0 101
>
> c=IN IP4 172.25.4.200
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=ptime:20
>
>
>
> 030593: .Dec 20 17:38:18.550: %VOICE_IEC-3-GW: SIP: Internal Error (1xx
> wait timeout): IEC=1.1.129.7.59.0 on callID 3480 GUID=
> 365A43AFC64411E69107D7209D491D8F
>
> 030594: .Dec 20 17:38:18.552: voice_reg_get_phone_index_from_contact:
> Trying to match key 2004
>
> 030595: .Dec 20 17:38:18.552: voice_reg_get_phone_index_from_contact:key
> MATCHED!!
>
>
>
> !!!### We receive the first of three trying responses at 17:39:31, almost
> 1 full minute from the last invite message at 17:38:17
>
> !!! the called phone rings at this point with a "phantom" call. and I
> picked up the handset
>
>
>
> 030596: .Dec 20 17:39:31.723: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>
> Received:
>
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
>
> From: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200>;tag=
> 5E6EC01-2547
>
> To: <sip:2004 at 172.25.4.26>
>
> Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
>
> Session-ID: 0ca99dec00105000a0006899cd85aa57;remote=
> e3d1d5b9cf5d5821bc81fbad4de46749
>
> Date: Tue, 20 Dec 2016 23:39:30 GMT
>
> CSeq: 101 INVITE
>
> Server: Cisco-CP7821/11.0.1
>
> Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060;
> transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
>
> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,
> SUBSCRIBE,INFO
>
> Supported: replaces,join,sdp-anat,norefersub,resource-priority,
> X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
>
> Allow-Events: kpml,dialog
>
> Content-Length: 0
>
>
>
>
>
> 030597: .Dec 20 17:39:31.761: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>
> Received:
>
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
>
> From: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200>;tag=
> 5E6EC01-2547
>
> To: <sip:2004 at 172.25.4.26>
>
> Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
>
> Session-ID: 0ca99dec00105000a0006899cd85aa57;remote=
> e3d1d5b9cf5d5821bc81fbad4de46749
>
> Date: Tue, 20 Dec 2016 23:39:30 GMT
>
> CSeq: 101 INVITE
>
> Server: Cisco-CP7821/11.0.1
>
> Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060;
> transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
>
> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,
> SUBSCRIBE,INFO
>
> Supported: replaces,join,sdp-anat,norefersub,resource-priority,
> X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
>
> Allow-Events: kpml,dialog
>
> Content-Length: 0
>
>
>
>
>
> 030599: .Dec 20 17:39:31.764: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>
> Received:
>
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
>
> From: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200>;tag=
> 5E6EC01-2547
>
> To: <sip:2004 at 172.25.4.26>
>
> Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
>
> Session-ID: 0ca99dec00105000a0006899cd85aa57;remote=
> e3d1d5b9cf5d5821bc81fbad4de46749
>
> Date: Tue, 20 Dec 2016 23:39:30 GMT
>
> CSeq: 101 INVITE
>
> Server: Cisco-CP7821/11.0.1
>
> Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060;
> transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
>
> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,
> SUBSCRIBE,INFO
>
> Supported: replaces,join,sdp-anat,norefersub,resource-priority,
> X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
>
> Allow-Events: kpml,dialog
>
> Content-Length: 0
>
>
>
>
>
> 030600: .Dec 20 17:39:31.818: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>
> Received:
>
> SIP/2.0 180 Ringing
>
> Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
>
> From: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200>;tag=
> 5E6EC01-2547
>
> To: <sip:2004 at 172.25.4.26>;tag=6899cd85aa57010c52826a47-3bf1086e
>
> Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
>
> Session-ID: 0ca99dec00105000a0006899cd85aa57;remote=
> e3d1d5b9cf5d5821bc81fbad4de46749
>
> Date: Tue, 20 Dec 2016 23:39:30 GMT
>
> CSeq: 101 INVITE
>
> Server: Cisco-CP7821/11.0.1
>
> Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060;
> transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
>
> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,
> SUBSCRIBE,INFO
>
> Remote-Party-ID: "Candace Flynn" <sip:2004 at 172.25.4.200>;party=
> called;id-type=subscriber;privacy=off;screen=yes
>
> Supported: replaces,join,sdp-anat,norefersub,resource-priority,
> X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
>
> Allow-Events: kpml,dialog
>
> Content-Length: 0
>
>
>
> !!!! ### phones sends back OK message at 17:41:31, almost 2 minutes after
> it sent back the 180 Ringing and I picked up handset
>
>
>
> 030603: .Dec 20 17:41:31.828: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>
> Received:
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
>
> From: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200>;tag=
> 5E6EC01-2547
>
> To: <sip:2004 at 172.25.4.26>;tag=6899cd85aa57010c52826a47-3bf1086e
>
> Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
>
> Session-ID: 0ca99dec00105000a0006899cd85aa57;remote=
> e3d1d5b9cf5d5821bc81fbad4de46749
>
> Date: Tue, 20 Dec 2016 23:41:30 GMT
>
> CSeq: 101 INVITE
>
> Server: Cisco-CP7821/11.0.1
>
> Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060;
> transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
>
> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,
> SUBSCRIBE,INFO
>
> Remote-Party-ID: "Candace Flynn" <sip:2004 at 172.25.4.200>;party=
> called;id-type=subscriber;privacy=off;screen=yes
>
> Supported: replaces,join,sdp-anat,norefersub,resource-priority,
> X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
>
> Allow-Events: kpml,dialog
>
> Content-Length: 217
>
> Content-Type: application/sdp
>
> Content-Disposition: session;handling=optional
>
>
>
> v=0
>
> o=Cisco-SIPUA 16080 0 IN IP4 172.25.4.26
>
> s=SIP Call
>
> t=0 0
>
> m=audio 28086 RTP/AVP 0 101
>
> c=IN IP4 172.25.4.26
>
> b=TIAS:64000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
> a=sendrecv
>
>
>
> 030604: .Dec 20 17:43:23.905: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>
> Received:
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
>
> From: "+12025552001 <(202)%20555-2001>" <sip:2001 at 172.25.4.200>;tag=
> 5E6EC01-2547
>
> To: <sip:2004 at 172.25.4.26>;tag=6899cd85aa57010c52826a47-3bf1086e
>
> Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
>
> Session-ID: 0ca99dec00105000a0006899cd85aa57;remote=
> e3d1d5b9cf5d5821bc81fbad4de46749
>
> Date: Tue, 20 Dec 2016 23:41:30 GMT
>
> CSeq: 101 INVITE
>
> Server: Cisco-CP7821/11.0.1
>
> Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060;
> transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
>
> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,
> SUBSCRIBE,INFO
>
> Remote-Party-ID: "Candace Flynn" <sip:2004 at 172.25.4.200>;party=
> called;id-type=subscriber;privacy=off;screen=yes
>
> Supported: replaces,join,sdp-anat,norefersub,resource-priority,
> X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
>
> Allow-Events: kpml,dialog
>
> Content-Length: 217
>
> Content-Type: application/sdp
>
> Content-Disposition: session;handling=optional
>
>
>
> v=0
>
> o=Cisco-SIPUA 16080 0 IN IP4 172.25.4.26
>
> s=SIP Call
>
> t=0 0
>
> m=audio 28086 RTP/AVP 0 101
>
> c=IN IP4 172.25.4.26
>
> b=TIAS:64000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
> a=sendrecv
>
>
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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