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870 Phones Part 2


AG1
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I have a Nextiva SIP trunk, I set my dial plan to dial that trunk (9*) when 9 is pushed then the number

 

This works on the Snom 821, Snom 720 and Grandstream GXP 2200 I can select the Nextiva trunk to dial out.

 

 

I also have a Sangoma NBE gateway that is the default if a 9 is not dialed. You can dial out on the 821,720, 2200 and 870 phones I have in the building.

 

 

The 870 phones will not dial out on the Nextiva trunk when you dial 9 then the number. The message that comes back is "you must first dial a 1 when calling this number" 91406209xxxx is what I am dialing

 

 

I have one standard dial plan set up in my PBX

 

101- Nextiva - 9*

102 - NBE - 911

103 - NBE - *

104 - NBE - 411

105 - NBE - 811

106 - NBE - ^([2-9][0-9]{6})@.*

107 - NBE - ^([0-9]{10})@.*

108 - NBE - ^(1[0-9]{10})@.*

 

All of the phones are assigned to this dial plan but the 870s will not dial out on my Nextiva trunk

 

 

Any ideas.......................I am probably just going to sell the 5 - 870 phones and get the GXP2200 instead since this is phone related.

 

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INVITE sip:914062094291@192.168.100.151;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport

From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg

To: <sip:914062094291@192.168.100.151;user=phone>

Call-ID: 57ca55528438-mg2rh2cjxzgr

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1

X-Serialnumber: 0004134150E1

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom870/8.7.3.19

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons-snom870

Content-Type: application/sdp

Content-Length: 517

 

v=0

o=root 974376235 974376235 IN IP4 192.168.100.93

s=call

c=IN IP4 192.168.100.93

t=0 0

m=audio 55204 RTP/AVP 9 0 8 99 108 18 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:sX/0lap2QdSW3yfqJoTZSpNIji/HGcd9jq4vyQfE

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:99 G726-32/8000

a=rtpmap:108 AAL2-G726-32/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

a=sendrecv [8] 2013/10/09 15:22:11: Packet authenticated by transport layer [8] 2013/10/09 15:22:11: Allocating for call port 389, SIP call id 57ca55528438-mg2rh2cjxzgr [8] 2013/10/09 15:22:11: Could not find a trunk (3 trunks) [5] 2013/10/09 15:22:11: SIP Tx tls:192.168.100.93:3372: SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport=3372

From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg

To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d

Call-ID: 57ca55528438-mg2rh2cjxzgr

CSeq: 1 INVITE

Content-Length: 0

[8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 6 (mapped to 9) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 3 (mapped to 8) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 7 (mapped to 18) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 5 (mapped to 99) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 10 (mapped to 108) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:11: Set packet length to 20 [6] 2013/10/09 15:22:11: Call-leg 389: Sending RTP for 57ca55528438-mg2rh2cjxzgr to 192.168.100.93:55204, codec not set yet [8] 2013/10/09 15:22:11: Incoming call: Request URI sip:914062094291@192.168.100.151;user=phone, To is <sip:914062094291@192.168.100.151;user=phone> [8] 2013/10/09 15:22:11: Call from an user 2203 [8] 2013/10/09 15:22:11: To is <sip:914062094291@192.168.100.151;user=phone>, user 0, domain 2 [8] 2013/10/09 15:22:11: From user 2203 [8] 2013/10/09 15:22:11: Set the To domain based on From user 2203@192.168.100.151 [8] 2013/10/09 15:22:11: Call state for call object 337: idle [7] 2013/10/09 15:22:11: Call port 389: Set codecs to "" preference count 3 [7] 2013/10/09 15:22:11: Skipping pattern match because CO-line is not available for trunk Nextiva [5] 2013/10/09 15:22:11: Dialplan "Standard Plan": Match 914062094291@192.168.100.151 to sip:914062094291@192.168.100.151:5066;user=phone on trunk Netborder Express [5] 2013/10/09 15:22:11: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:11: Allocating for call port 390, SIP call id 50aa13fa@pbx [7] 2013/10/09 15:22:11: Call port 390: Set codecs to "0" preference count 2 [5] 2013/10/09 15:22:11: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:11: Call port 390: state code from 0 to 100 [5] 2013/10/09 15:22:11: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:11: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:11: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:11: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:11: Play audio_moh/noise.wav, caching true [5] 2013/10/09 15:22:11: SIP Tx udp:192.168.100.151:5066: INVITE sip:914062094291@192.168.100.151:5066;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721

To: <sip:914062094291@192.168.100.151;user=phone>

Call-ID: 50aa13fa@pbx

CSeq: 29245 INVITE

Max-Forwards: 70

Contact: <sip:anonymous@192.168.100.151:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/5.0.10

P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066>

Privacy: id

Content-Type: application/sdp

Content-Length: 243

 

v=0

o=- 154740586 154740586 IN IP4 192.168.100.151

s=-

c=IN IP4 192.168.100.151

t=0 0

m=audio 60494 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv [5] 2013/10/09 15:22:11: SIP Rx udp:192.168.100.151:5066: SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport=5060

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721

To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574

Call-ID: 50aa13fa@pbx

CSeq: 29245 INVITE

Content-Length: 0

Server: Netborder Express Gateway/4.3.13

Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp>

[8] 2013/10/09 15:22:11: Call port 389: state code from 0 to 183 [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 6 (mapped to 9) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 3 (mapped to 8) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 7 (mapped to 18) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 5 (mapped to 99) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 10 (mapped to 108) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:11: Set packet length to 20 [6] 2013/10/09 15:22:11: Call-leg 389: Codec PCMU/8000 is chosen for call id 57ca55528438-mg2rh2cjxzgr [5] 2013/10/09 15:22:11: set codec: codec PCMU/8000 is set to call-leg 389 [5] 2013/10/09 15:22:11: SIP Tx tls:192.168.100.93:3372: SIP/2.0 183 Session Progress

Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport=3372

From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg

To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d

Call-ID: 57ca55528438-mg2rh2cjxzgr

CSeq: 1 INVITE

Contact: <sip:2203@192.168.100.151:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/5.0.10

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 386

 

v=0

o=- 639170658 639170658 IN IP4 192.168.100.151

s=-

c=IN IP4 192.168.100.151

t=0 0

m=audio 49472 RTP/AVP 0 18 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+mUYvsDtufVA2IS4U3Q3DZB12+ZzLjFbdKTFyIiv

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv [5] 2013/10/09 15:22:11: SIP Rx tls:192.168.100.93:3372: PRACK sip:2203@192.168.100.151:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-az5g906p0x93;rport

From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg

To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d

Call-ID: 57ca55528438-mg2rh2cjxzgr

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1

RAck: 1 1 INVITE

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info

Proxy-Require: buttons-snom870

Content-Length: 0

[8] 2013/10/09 15:22:11: Packet authenticated by transport layer [5] 2013/10/09 15:22:11: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-az5g906p0x93;rport=3372

From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg

To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d

Call-ID: 57ca55528438-mg2rh2cjxzgr

CSeq: 2 PRACK

Contact: <sip:2203@192.168.100.151:5061;transport=tls>

User-Agent: snomONE/5.0.10

Content-Length: 0

[8] 2013/10/09 15:22:11: Packet authenticated by transport layer [8] 2013/10/09 15:22:12: Last message repeated 3 times [8] 2013/10/09 15:22:12: SMTP: Connect to 74.125.140.108:25 [8] 2013/10/09 15:22:13: SMTP: Received 220 mx.google.com ESMTP v22sm64056817yhn.12 - gsmtp [8] 2013/10/09 15:22:13: SMTP: Send EHLO localhost [8] 2013/10/09 15:22:13: Packet authenticated by transport layer [8] 2013/10/09 15:22:13: SMTP: Received 250-mx.google.com at your service, [216.228.51.194]

250-SIZE 35882577

250-8BITMIME

250-STARTTLS

250-ENHANCEDSTATUSCODES

250 CHUNKING [8] 2013/10/09 15:22:13: SMTP: Send STARTTLS [8] 2013/10/09 15:22:13: SMTP: Received 220 2.0.0 Ready to start TLS [8] 2013/10/09 15:22:13: SMTP: Send EHLO localhost [4] 2013/10/09 15:22:13: Certificate for Equifax Secure Certificate Authority not available [5] 2013/10/09 15:22:13: Certificate for smtp.gmail.com could not be verified against [8] 2013/10/09 15:22:13: Play audio_en/aa_no_answer.wav space10 audio_en/aa_receive_callback.wav audio_en/aa_leave_message.wav audio_en/aa_offer_cellphone.wav space50, caching false [8] 2013/10/09 15:22:13: Call port 379: state code from 200 to 200 [8] 2013/10/09 15:22:13: Trunk 4: Preparing for re-registration [8] 2013/10/09 15:22:13: Trunk Nextiva: Sending registration to 208.73.146.95 [8] 2013/10/09 15:22:13: Trunk 4: setup callback to send re-registration after 38 seconds [5] 2013/10/09 15:22:13: SMTP: Connection to 74.125.140.108:25 failed [8] 2013/10/09 15:22:14: Packet authenticated by transport layer [8] 2013/10/09 15:22:16: Last message repeated 4 times [5] 2013/10/09 15:22:16: SIP Rx udp:192.168.100.151:5066: SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport=5060

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721

To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574

Call-ID: 50aa13fa@pbx

CSeq: 29245 INVITE

Content-Length: 238

Content-Type: application/sdp

Server: Netborder Express Gateway/4.3.13

Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp>

 

v=0

o=Sangoma-Tech 1381353736 1381353785 IN IP4 192.168.100.151

s=SIP Call

c=IN IP4 192.168.100.151

t=0 0

m=audio 14574 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv [8] 2013/10/09 15:22:16: Call port 390: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:16: Call port 390: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:16: Set packet length to 20 [6] 2013/10/09 15:22:16: Call-leg 390: Codec PCMU/8000 is chosen for call id 50aa13fa@pbx [6] 2013/10/09 15:22:16: Call-leg 390: Sending RTP for 50aa13fa@pbx to 192.168.100.151:14574, codec PCMU/8000 [5] 2013/10/09 15:22:16: set codec: codec PCMU/8000 is set to call-leg 390 [5] 2013/10/09 15:22:16: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:16: Call state for call object 337: alerting [8] 2013/10/09 15:22:16: Call port 389: state code from 183 to 183 [8] 2013/10/09 15:22:16: Last message repeated 2 times [7] 2013/10/09 15:22:16: 57ca55528438-mg2rh2cjxzgr: RTP pass-through mode [7] 2013/10/09 15:22:16: 50aa13fa@pbx: RTP pass-through mode [5] 2013/10/09 15:22:16: SIP Rx udp:192.168.100.151:5066: SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport=5060

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721

To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574

Call-ID: 50aa13fa@pbx

CSeq: 29245 INVITE

Content-Length: 238

Content-Type: application/sdp

Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp>

Server: Netborder Express Gateway/4.3.13

 

v=0

o=Sangoma-Tech 1381353736 1381353786 IN IP4 192.168.100.151

s=SIP Call

c=IN IP4 192.168.100.151

t=0 0

m=audio 14574 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv [7] 2013/10/09 15:22:16: Call 50aa13fa@pbx: Clear last INVITE [8] 2013/10/09 15:22:16: Call port 390: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:16: Call port 390: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:16: Set packet length to 20 [5] 2013/10/09 15:22:16: SIP Tx udp:192.168.100.151:5066: ACK sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-b0002f2d1f9910b246232d9f43d1c0b0;rport

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721

To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574

Call-ID: 50aa13fa@pbx

CSeq: 29245 ACK

Max-Forwards: 70

Contact: <sip:anonymous@192.168.100.151:5060;transport=udp>

P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066>

Privacy: id

Content-Length: 0

[7] 2013/10/09 15:22:16: Determine pass-through mode after receiving response [8] 2013/10/09 15:22:16: Call state for call object 337: connected [5] 2013/10/09 15:22:16: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:16: Call port 390: state code from 100 to 200 [8] 2013/10/09 15:22:16: Call port 389: state code from 183 to 200 [5] 2013/10/09 15:22:16: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport=3372

From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg

To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d

Call-ID: 57ca55528438-mg2rh2cjxzgr

CSeq: 1 INVITE

Contact: <sip:2203@192.168.100.151:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/5.0.10

Content-Type: application/sdp

Content-Length: 386

 

v=0

o=- 639170658 639170658 IN IP4 192.168.100.151

s=-

c=IN IP4 192.168.100.151

t=0 0

m=audio 49472 RTP/AVP 0 18 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+mUYvsDtufVA2IS4U3Q3DZB12+ZzLjFbdKTFyIiv

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv [5] 2013/10/09 15:22:16: SIP Rx tls:192.168.100.93:3372: ACK sip:2203@192.168.100.151:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-3ojncdnhtw1j;rport

From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg

To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d

Call-ID: 57ca55528438-mg2rh2cjxzgr

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1

Proxy-Require: buttons-snom870

Content-Length: 0

[8] 2013/10/09 15:22:16: Packet authenticated by transport layer [8] 2013/10/09 15:22:18: Last message repeated 6 times [8] 2013/10/09 15:22:18: SMTP: Connect to 74.125.140.108:25 [8] 2013/10/09 15:22:19: SMTP: Received 421 4.4.5 Server busy, try again later. (mx.google.com) s21sm64052493yhk.9 - gsmtp [5] 2013/10/09 15:22:19: SMTP Server returned 421 [8] 2013/10/09 15:22:19: Packet authenticated by transport layer [8] 2013/10/09 15:22:22: Last message repeated 9 times [8] 2013/10/09 15:22:22: rtp_hangup: call port 387, too early to disconnect [8] 2013/10/09 15:22:22: rtp_hangup: call port 388, too early to disconnect [8] 2013/10/09 15:22:23: Packet authenticated by transport layer [8] 2013/10/09 15:22:26: Last message repeated 6 times [8] 2013/10/09 15:22:26: Play audio_en/aa_no_answer.wav space10 audio_en/aa_receive_callback.wav audio_en/aa_leave_message.wav audio_en/aa_offer_cellphone.wav space50, caching false [8] 2013/10/09 15:22:26: Call port 379: state code from 200 to 200 [8] 2013/10/09 15:22:29: Packet authenticated by transport layer [8] 2013/10/09 15:22:35: Last message repeated 12 times [5] 2013/10/09 15:22:35: Identify trunk (IP address/port and domain match) 3 [8] 2013/10/09 15:22:36: Packet authenticated by transport layer [5] 2013/10/09 15:22:36: SIP Rx tls:192.168.100.93:3372: BYE sip:2203@192.168.100.151:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-u7zgtvycra3f;rport

From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg

To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d

Call-ID: 57ca55528438-mg2rh2cjxzgr

CSeq: 3 BYE

Max-Forwards: 70

Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1

User-Agent: snom870/8.7.3.19

RTP-RxStat: Total_Rx_Pkts=1242,Rx_Pkts=1227,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=1229,Tx_Pkts=1229,Remote_Tx_Pkts=6

Proxy-Require: buttons-snom870

Content-Length: 0

[8] 2013/10/09 15:22:36: Packet authenticated by transport layer [5] 2013/10/09 15:22:36: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-u7zgtvycra3f;rport=3372

From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg

To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d

Call-ID: 57ca55528438-mg2rh2cjxzgr

CSeq: 3 BYE

Contact: <sip:2203@192.168.100.151:5061;transport=tls>

User-Agent: snomONE/5.0.10

Content-Length: 0

[7] 2013/10/09 15:22:36: 50aa13fa@pbx: Media-aware pass-through mode [8] 2013/10/09 15:22:36: Clearing call port 389, SIP call id 57ca55528438-mg2rh2cjxzgr [8] 2013/10/09 15:22:36: Call port 390: state code from 200 to 486 [5] 2013/10/09 15:22:36: SIP Tx udp:192.168.100.151:5066: BYE sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-3f99471547e748b3382e3a011de7b126;rport

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721

To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574

Call-ID: 50aa13fa@pbx

CSeq: 29246 BYE

Max-Forwards: 70

Contact: <sip:anonymous@192.168.100.151:5060;transport=udp>

P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066>

Privacy: id

Content-Length: 0

[5] 2013/10/09 15:22:36: SIP Rx udp:192.168.100.151:5066: SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-3f99471547e748b3382e3a011de7b126;rport=5060

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721

To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574

Call-ID: 50aa13fa@pbx

CSeq: 29246 BYE

Content-Length: 0

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Yes I have 4 CO lines configured on the phone, I deleted them on one of the phones and now the phone will call out on the Nextiva trunk.

 

 

Now I have a question about BLF, the reason I had co lines configured was so the operator and 4 other people could see lines ringing into the building and answer them even though they are not in the hunt group.

 

If I use BLF to configure a button to monitor incoming calls would I put "3000" in the parameters box since that is my hunt group?

 

If not how would I make 4 buttons you can monitor and answer on the virtual buttons?

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