# Pickup detection problem **URL:** \ **Category:** Asterisk Support\ **Created:** [September 19, 2010, 3:25pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606 "2010-09-19T15:25:48Z")\ **Posts on this page:** 14\ **Page:** 1
**Author:** ![underattack](https://avatars.discourse-cdn.com/v4/letter/u/ecc23a/32.png) [@underattack](https://community.asterisk.org/u/underattack)\ **Post date:** [September 19, 2010, 3:25pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/1 "2010-09-19T15:25:48Z")
Hi, I am running Asterisk 1.6.2.13 I have a problem on one phone (only), which does not detect a call pickup, this phone is a Nortel 1535. I can give a ring without any problem to this phone, but when I call a number from this phone, it does not detect that the other side picks up the phone, and it keeps ringing. Weird thing is that I can see RTP traffic between the 2 phones, though the Nortel is still ringing. I have tried several parameters without any success : dtmfmode = rfc2833 rfc2833compensate = yes progressinband = yes Any idea ? Thanks very for your help as I run out of ideas. Fabien ---
**Author:** ![underattack](https://avatars.discourse-cdn.com/v4/letter/u/ecc23a/32.png) [@underattack](https://community.asterisk.org/u/underattack)\ **Post date:** [September 19, 2010, 3:42pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/2 "2010-09-19T15:42:21Z")
just on thing that might help troubleshooting this problem : If I give a ring from the Nortel to a phone A, when A picks-up , I can see “call connected” on the Nortel, but the Nortel is still ringing. Then if on the Nortel, I put the communication on-hold, and resume it, it stops ringing and I can talk to phone A and hear it as well…that’s crazy ! Cheers, Fabien ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [September 20, 2010, 5:57am UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/3 "2010-09-20T05:57:23Z")
Fabian, I think that the problem could be on the SIP messages. Could you please enable SIP DEBUG in asterisk and check if Asterisk is sending the “200 OK” to the Nortel when the other phones answers the call. ---
**Author:** ![underattack](https://avatars.discourse-cdn.com/v4/letter/u/ecc23a/32.png) [@underattack](https://community.asterisk.org/u/underattack)\ **Post date:** [September 20, 2010, 9:33pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/4 "2010-09-20T21:33:56Z")
Hi, Yes apparently I have a SIP 200 OK : 192.168.1.18 is the Nortel Phone form which I give the call with extension 121. 192.168.1.72 is the phone receiving the call with extension 102 192.168.1.15 is the Asterisk server. (I replaced my domain name per “my.domain” in the logs) This is the end of the debug logs: \<— SIP read from UDP:192.168.1.72:50541 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK65f69116;rport=5163 To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975 From: “Cambridge” sip:121@my.domain;tag=as693d29d7 Call-ID: [4995bf7656e2875e347770af0109c756@my.domain](mailto:4995bf7656e2875e347770af0109c756@my.domain) CSeq: 102 INVITE Contact: sip:102@192.168.1.72:50541;transport=udp Server: Sipdroid/1.5.7 beta/HTC Hero Content-Length: 189 Content-Type: application/sdp v=0 o=102@my.domain\> 0 0 INIP4 192.168.1.72 s=Session SIP/SDP c=IN IP4 192.168.1.72 t=0 0 m=audio 21000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 \<-------------\> — (10 headers 8 lines) — list\_route: hop: sip:102@192.168.1.72:50541;transport=udp set\_destination: Parsing sip:102@192.168.1.72:50541;transport=udp for address/port to send to set\_destination: set destination to 192.168.1.72, port 50541 Transmitting (no NAT) to 192.168.1.72:50541: ACK sip:102@192.168.1.72:50541;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK6f399895;rport Max-Forwards: 70 From: “Cambridge” sip:121@my.domain;tag=as693d29d7 To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975 Contact: sip:121@192.168.1.15:5163 Call-ID: [4995bf7656e2875e347770af0109c756@my.domain](mailto:4995bf7656e2875e347770af0109c756@my.domain) CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 * * * Audio is at 192.168.1.15 port 10086 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP \<— Reliably Transmitting (no NAT) to 192.168.1.18:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603 To: sip:102@my.domain;tag=as65ce952f Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603 CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:102@192.168.1.15:5163 Content-Type: application/sdp Content-Length: 261 v=0 o=root 1949615126 1949615127 IN IP4 192.168.1.15 s=Asterisk PBX 1.6.2.13 c=IN IP4 192.168.1.15 t=0 0 m=audio 10086 RTP/AVP 0 99 a=rtpmap:0 PCMU/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv \<------------\> Thanks for your help, Fabien ---
**Author:** ![underattack](https://avatars.discourse-cdn.com/v4/letter/u/ecc23a/32.png) [@underattack](https://community.asterisk.org/u/underattack)\ **Post date:** [September 20, 2010, 10:40pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/5 "2010-09-20T22:40:08Z")
Hi again, Also I did a tcpdump, and I can see the following: Nortel -\> asterisk INVITE Phone -\> Asterisk Status 180 Ringing Asterisk -\> Nortel Status 180 Ringing Asterisk -\> Nortel status 183 Session Progress Then RTP traffic is sent from Nortel to the phone via Asterisk (in this way only): Then RTP traffic is also sent from the phone to the Nortel via Asterisk And later, the phone sends a SIP OK 200 but the Nortel still rings… Thanks, Fabien ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [September 21, 2010, 9:59am UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/6 "2010-09-21T09:59:26Z")
The problem is with the Nortel. ---
**Author:** ![underattack](https://avatars.discourse-cdn.com/v4/letter/u/ecc23a/32.png) [@underattack](https://community.asterisk.org/u/underattack)\ **Post date:** [September 21, 2010, 10:51am UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/7 "2010-09-21T10:51:11Z")
Hi David, Thing is that the problem does not occur with all the phones. It only happen when I use a SIP trunk, or to a few phones. Is there any particular setting in Asterisk that could help solving this issue ? (like wait for 200 SIP before sending RTP traffic) Thanks, Fabien ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [September 21, 2010, 11:11am UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/8 "2010-09-21T11:11:01Z")
Sorry. I am being confused about the directions here. The problem, I think, is that you have said rings, when you actually mean receives ringback tone. Also, your tcpdump summary is missing the INVITE from Asterisk to the Phone, which made me think that the 180 and 183 where associated with the original INVITE. Asterisk should not be initiating early media unless the dialplan tells it to, or the called party requests it. In the latest versions, it shouldn’t initiate it unless the dialplan explicitly requests it (by using the Progress() application). Asterisk should be sending 200 OK when it receives 200 OK from the phone. It appears to be doing that: \<— Reliably Transmitting (no NAT) to 192.168.1.18:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18 At that point it should cease sending any internally generated early media. You need the complete debug log to see whether Asterisk was trying to source ringback tone, or whether it is just passing it through. There are options to disable the use of early media. ---
**Author:** ![underattack](https://avatars.discourse-cdn.com/v4/letter/u/ecc23a/32.png) [@underattack](https://community.asterisk.org/u/underattack)\ **Post date:** [September 21, 2010, 10:45pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/9 "2010-09-21T22:45:13Z")
Hi, Here is a trace : ## SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bd9-e961bad;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603 To: sip:102@my.domain;tag=as701ad038 Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603 ## SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603 To: sip:102@my.domain Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603 ## SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK65f69116;rport=5163 To: sip:102@192.168.1.72:50541;transport=udp From: “home” sip:121@my.domain;tag=as693d29d7 Call-ID: [4995bf7656e2875e347770af0109c756@my.domain](mailto:4995bf7656e2875e347770af0109c756@my.domain) ## SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK65f69116;rport=5163 To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975 From: “home” sip:121@my.domain;tag=as693d29d7 Call-ID: [4995bf7656e2875e347770af0109c756@my.domain](mailto:4995bf7656e2875e347770af0109c756@my.domain) ## SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603 To: sip:102@my.domain;tag=as65ce952f Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603 ## SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603 To: sip:102@my.domain;tag=as65ce952f Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603 ## SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK65f69116;rport=5163 To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975 From: “home” sip:121@my.domain;tag=as693d29d7 Call-ID: [4995bf7656e2875e347770af0109c756@my.domain](mailto:4995bf7656e2875e347770af0109c756@my.domain) ## SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-603-177bfb-44f9b6a3;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603 To: sip:102@my.domain;tag=as65ce952f Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603 ## SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-615-17c433-a626eb1;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2f5f98-1201a8c0-13c4-45026-603-4f9b78e7-603 To: sip:102@my.domain;tag=as65ce952f Call-ID: 303b80-1201a8c0-13c4-45026-603-2aa658b9-603 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK0ba736bc;rport=5163 To: sip:102@192.168.1.72:50541;transport=udp;tag=4d439f22e1ddd975 From: “home” sip:121@my.domain;tag=as693d29d7 Call-ID: [4995bf7656e2875e347770af0109c756@my.domain](mailto:4995bf7656e2875e347770af0109c756@my.domain) Thanks, regards, Fabien ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [September 22, 2010, 9:29am UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/10 "2010-09-22T09:29:40Z")
That’s a very incomplete trace. There is only one side. The SIP headers are incomplete and there is no SDP payload. There is no Asterisk debug output showing what it is doing with the messages. It’s not worth my time looking at it in any detail. ---
**Author:** ![underattack](https://avatars.discourse-cdn.com/v4/letter/u/ecc23a/32.png) [@underattack](https://community.asterisk.org/u/underattack)\ **Post date:** [September 22, 2010, 10:16pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/11 "2010-09-22T22:16:43Z")
Hi David, I tried to cut the last trace, sorry about that. Here is a complete trace : Nortel phone at 192.168.1.18 gives a ring to phone at 192.168.1.72. Asterisk is at 192.168.1.15. When the phone picks up, the Nortel phone keeps on ringing, then the phone hangs up. Weird thing is I can see the first SIP 200 OK coming from the Nortel. \<— SIP read from UDP:192.168.1.18:5060 —\> INVITE sip:102@my.domain SIP/2.0 From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d To: sip:102@my.domain Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba50-3292b1a4 Max-Forwards: 70 Supported: replaces User-Agent: Nortel IP Phone 1535 (0.2.50.0905) Contact: sip:121@192.168.1.18:5060 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Content-Type: application/sdp Content-Length: 400 v=0 o=LGEIPP 5762 5762 IN IP4 192.168.1.18 s=SIP Call c=IN IP4 192.168.1.18 b=CT:160 t=0 0 m=audio 23000 RTP/AVP 0 8 18 4 95 99 111 c=IN IP4 192.168.1.18 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:95 AMR/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=sendrecv \<-------------\> — (13 headers 19 lines) — Sending to 192.168.1.18 : 5060 (NAT) Using INVITE request as basis request - 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d Found peer ‘121’ for ‘121’ from 192.168.1.18:5060 \<— Reliably Transmitting (no NAT) to 192.168.1.18:5060 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba50-3292b1a4;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d To: sip:102@my.domain;tag=as2885dfe9 Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="65123c99" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog ‘304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d’ in 32000 ms (Method: INVITE) \<— SIP read from UDP:192.168.1.18:5060 —\> ACK sip:102@my.domain SIP/2.0 From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d To: sip:102@my.domain;tag=as2885dfe9 Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba50-3292b1a4 Max-Forwards: 70 User-Agent: Nortel IP Phone 1535 (0.2.50.0905) Contact: sip:121@192.168.1.18:5060 Content-Length: 0 \<-------------\> — (10 headers 0 lines) — \<— SIP read from UDP:192.168.1.18:5060 —\> INVITE sip:102@my.domain SIP/2.0 From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d To: sip:102@my.domain Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a Max-Forwards: 70 Supported: replaces User-Agent: Nortel IP Phone 1535 (0.2.50.0905) Contact: sip:121@192.168.1.18:5060 Authorization: Digest username=“121”,realm=“asterisk”,nonce=“65123c99”,uri=“sip:102@my.domain”,response=“d43d93964059735a8a1afeab3c96bbb3”,algorithm=MD5 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Content-Type: application/sdp Content-Length: 400 v=0 o=LGEIPP 5762 5762 IN IP4 192.168.1.18 s=SIP Call c=IN IP4 192.168.1.18 b=CT:160 t=0 0 m=audio 23000 RTP/AVP 0 8 18 4 95 99 111 c=IN IP4 192.168.1.18 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:95 AMR/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=sendrecv \<-------------\> — (14 headers 19 lines) — Sending to 192.168.1.18 : 5060 (no NAT) Using INVITE request as basis request - 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d Found peer ‘121’ for ‘121’ from 192.168.1.18:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 95 Found RTP audio format 99 Found RTP audio format 111 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format AMR for ID 95 Found audio description format telephone-event for ID 99 Found audio description format X-nt-inforeq for ID 111 Capabilities: us - 0x380004 (ulaw|h263|h263p|h264), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.18:23000 Peer doesn’t provide video Looking for 102 in DLPN\_DialPlan1 (domain my.domain) list\_route: hop: sip:121@192.168.1.18:5060 \<— Transmitting (no NAT) to 192.168.1.18:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d To: sip:102@my.domain Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:102@192.168.1.15:5163 Content-Length: 0 \<------------\> Audio is at 192.168.1.15 port 10078 Video is at 192.168.1.15 port 10070 Adding codec 0x4 (ulaw) to SDP Adding video codec 0x80000 (h263) to SDP Adding video codec 0x100000 (h263p) to SDP Adding video codec 0x200000 (h264) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.72:34702: INVITE sip:102@192.168.1.72:34702;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK3fd058b0;rport Max-Forwards: 70 From: “home” sip:121@my.domain;tag=as53a84d81 To: sip:102@192.168.1.72:34702;transport=udp Contact: sip:121@192.168.1.15:5163 Call-ID: [64dc54dc7366e87d6d06cef8780b5cfa@my.domain](mailto:64dc54dc7366e87d6d06cef8780b5cfa@my.domain) CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 22 Sep 2010 21:50:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 393 v=0 o=root 234067187 234067187 IN IP4 192.168.1.15 s=Asterisk PBX 1.6.2.13 c=IN IP4 192.168.1.15 b=CT:384 t=0 0 m=audio 10078 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 10070 RTP/AVP 34 98 99 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv * * * \<— SIP read from UDP:192.168.1.72:34702 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK3fd058b0;rport=5163 To: sip:102@192.168.1.72:34702;transport=udp From: “home” sip:121@my.domain;tag=as53a84d81 Call-ID: [64dc54dc7366e87d6d06cef8780b5cfa@my.domain](mailto:64dc54dc7366e87d6d06cef8780b5cfa@my.domain) CSeq: 102 INVITE Server: Sipdroid/1.5.7 beta/HTC Hero Content-Length: 0 * * * \<— SIP read from UDP:192.168.1.72:34702 —\> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK3fd058b0;rport=5163 To: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f From: “home” sip:121@my.domain;tag=as53a84d81 Call-ID: [64dc54dc7366e87d6d06cef8780b5cfa@my.domain](mailto:64dc54dc7366e87d6d06cef8780b5cfa@my.domain) CSeq: 102 INVITE Server: Sipdroid/1.5.7 beta/HTC Hero Content-Length: 189 Content-Type: application/sdp v=0 o=102@my.domain 0 0 IN IP4 192.168.1.72 s=Session SIP/SDP c=IN IP4 192.168.1.72 t=0 0 m=audio 21000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 \<-------------\> — (9 headers 8 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x380004 (ulaw|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.72:21000 Peer doesn’t provide video \<— Transmitting (no NAT) to 192.168.1.18:5060 —\> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d To: sip:102@my.domain;tag=as4162285c Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:102@192.168.1.15:5163 Content-Length: 0 \<------------\> Audio is at 192.168.1.15 port 10076 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP \<— Transmitting (no NAT) to 192.168.1.18:5060 —\> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d To: sip:102@my.domain;tag=as4162285c Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:102@192.168.1.15:5163 Content-Type: application/sdp Content-Length: 259 v=0 o=root 954720937 954720937 IN IP4 192.168.1.15 s=Asterisk PBX 1.6.2.13 c=IN IP4 192.168.1.15 t=0 0 m=audio 10076 RTP/AVP 0 99 a=rtpmap:0 PCMU/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv * * * \<— SIP read from UDP:192.168.1.72:34702 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK3fd058b0;rport=5163 To: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f From: “home” sip:121@my.domain;tag=as53a84d81 Call-ID: [64dc54dc7366e87d6d06cef8780b5cfa@my.domain](mailto:64dc54dc7366e87d6d06cef8780b5cfa@my.domain) CSeq: 102 INVITE Contact: sip:102@192.168.1.72:34702;transport=udp Server: Sipdroid/1.5.7 beta/HTC Hero Content-Length: 189 Content-Type: application/sdp v=0 o=102@my.domain 0 0 IN IP4 192.168.1.72 s=Session SIP/SDP c=IN IP4 192.168.1.72 t=0 0 m=audio 21000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 \<-------------\> — (10 headers 8 lines) — list\_route: hop: sip:102@192.168.1.72:34702;transport=udp set\_destination: Parsing sip:102@192.168.1.72:34702;transport=udp for address/port to send to set\_destination: set destination to 192.168.1.72, port 34702 Transmitting (no NAT) to 192.168.1.72:34702: ACK sip:102@192.168.1.72:34702;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK7b855d14;rport Max-Forwards: 70 From: “home” sip:121@my.domain;tag=as53a84d81 To: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f Contact: sip:121@192.168.1.15:5163 Call-ID: [64dc54dc7366e87d6d06cef8780b5cfa@my.domain](mailto:64dc54dc7366e87d6d06cef8780b5cfa@my.domain) CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 * * * Audio is at 192.168.1.15 port 10076 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP \<— Reliably Transmitting (no NAT) to 192.168.1.18:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d1d-1e8ba67-1cef350a;received=192.168.1.18 From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d To: sip:102@my.domain;tag=as4162285c Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:102@192.168.1.15:5163 Content-Type: application/sdp Content-Length: 259 v=0 o=root 954720937 954720938 IN IP4 192.168.1.15 s=Asterisk PBX 1.6.2.13 c=IN IP4 192.168.1.15 t=0 0 m=audio 10076 RTP/AVP 0 99 a=rtpmap:0 PCMU/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv \<------------\> \<— SIP read from UDP:192.168.1.18:5060 —\> ACK sip:102@192.168.1.15:5163 SIP/2.0 From: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d To: sip:102@my.domain;tag=as4162285c Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 2 ACK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK-7d26-1e8df86-59f5a8b6 Max-Forwards: 70 User-Agent: Nortel IP Phone 1535 (0.2.50.0905) Contact: sip:121@192.168.1.18:5060 Authorization: Digest username=“121”,realm=“asterisk”,nonce=“65123c99”,uri=“sip:102@my.domain”,response=“d43d93964059735a8a1afeab3c96bbb3”,algorithm=MD5 Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Reliably Transmitting (no NAT) to 192.168.1.72:34702: OPTIONS sip:102@192.168.1.72:34702;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK32634603;rport Max-Forwards: 70 From: “asterisk” sip:asterisk@my.domain;tag=as11fdd8cb To: sip:102@192.168.1.72:34702;transport=udp Contact: sip:asterisk@192.168.1.15:5163 Call-ID: [576d57b62e7f8dfd37ca1790325bfd77@my.domain](mailto:576d57b62e7f8dfd37ca1790325bfd77@my.domain) CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 22 Sep 2010 21:50:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 * * * \<— SIP read from UDP:192.168.1.72:34702 —\> BYE sip:121@192.168.1.15:5163 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.72:34702;rport;branch=z9hG4bK39294 Max-Forwards: 70 To: “home” sip:121@my.domain;tag=as53a84d81 From: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f Call-ID: [64dc54dc7366e87d6d06cef8780b5cfa@my.domain](mailto:64dc54dc7366e87d6d06cef8780b5cfa@my.domain) CSeq: 1 BYE User-Agent: Sipdroid/1.5.7 beta/HTC Hero Content-Length: 0 \<-------------\> — (9 headers 0 lines) — Sending to 192.168.1.72 : 34702 (no NAT) \<— Transmitting (no NAT) to 192.168.1.72:34702 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.72:34702;branch=z9hG4bK39294;received=192.168.1.72;rport=34702 From: sip:102@192.168.1.72:34702;transport=udp;tag=9cb3773777cd3b0f To: “home” sip:121@my.domain;tag=as53a84d81 Call-ID: [64dc54dc7366e87d6d06cef8780b5cfa@my.domain](mailto:64dc54dc7366e87d6d06cef8780b5cfa@my.domain) CSeq: 1 BYE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog ‘304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d’ in 32000 ms (Method: ACK) set\_destination: Parsing sip:121@192.168.1.18:5060 for address/port to send to set\_destination: set destination to 192.168.1.18, port 5060 Reliably Transmitting (no NAT) to 192.168.1.18:5060: BYE sip:121@192.168.1.18:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK1aa94777;rport Max-Forwards: 70 From: sip:102@my.domain;tag=as4162285c To: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 102 BYE User-Agent: Asterisk PBX X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 * * * Really destroying SIP dialog '64dc54dc7366e87d6d06cef8780b5cfa@my.domain’ Method: BYE \<— SIP read from UDP:192.168.1.18:5060 —\> SIP/2.0 200 OK From: sip:102@my.domain;tag=as4162285c To: "121"sip:121@my.domain;tag=2fabf8-1201a8c0-13c4-45026-7d1d-616b605a-7d1d Call-ID: 304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d CSeq: 102 BYE Via: SIP/2.0/UDP 192.168.1.15:5163;rport=5163;branch=z9hG4bK1aa94777 Supported: replaces Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: Nortel IP Phone 1535 (0.2.50.0905) Content-Length: 0 \<-------------\> — (10 headers 0 lines) — SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog ‘304f70-1201a8c0-13c4-45026-7d1d-50249efc-7d1d’ Method: ACK Retransmitting #1 (no NAT) to 192.168.1.72:34702: OPTIONS sip:102@192.168.1.72:34702;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK32634603;rport Max-Forwards: 70 From: “asterisk” sip:asterisk@my.domain;tag=as11fdd8cb To: sip:102@192.168.1.72:34702;transport=udp Contact: sip:asterisk@192.168.1.15:5163 Call-ID: [576d57b62e7f8dfd37ca1790325bfd77@my.domain](mailto:576d57b62e7f8dfd37ca1790325bfd77@my.domain) CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 22 Sep 2010 21:50:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 * * * \<— SIP read from UDP:192.168.1.72:34702 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.15:5163;branch=z9hG4bK32634603;rport=5163 To: sip:102@192.168.1.72:34702;transport=udp From: “asterisk” sip:asterisk@my.domain;tag=as11fdd8cb Call-ID: [576d57b62e7f8dfd37ca1790325bfd77@my.domain](mailto:576d57b62e7f8dfd37ca1790325bfd77@my.domain) CSeq: 102 OPTIONS Contact: sip:102@192.168.1.72:34702;transport=udp Content-Length: 0 \<-------------\> — (8 headers 0 lines) — Really destroying SIP dialog '576d57b62e7f8dfd37ca1790325bfd77@my.domain’ Method: OPTIONS \<— SIP read from UDP:192.168.1.72:34702 —\> \<-------------\> Is it a issue with early media ? How can I remove it if it is the case ? Adding “progress …” in the Dial function ? Thanks for your help, Regards, Fabien ---
**Author:** ![underattack](https://avatars.discourse-cdn.com/v4/letter/u/ecc23a/32.png) [@underattack](https://community.asterisk.org/u/underattack)\ **Post date:** [September 22, 2010, 11:39pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/12 "2010-09-22T23:39:14Z")
Hi David, I think I found a solution (thanks to your comments). I have added “r” in the Dial function in extensions.conf (exten = 1-dial,1,Dial(${ARG1},r) and now the Nortel stops ringing when the other sides picks-up, and the communication works fine. From the documentation of the Dial function: r — indicate call ringing to the calling extension This option is used when your phone set doesn’t recognize the traditional call progress indications. The best example of this is when you don’t hear the “ringing” on the phone when you call another extension. With the ‘r’ option turned on, ring tone is passed as actual audio on the phone line, allowing you to hear the call progress. Is this something that can be a problem as the parameter that I modified apply for every phones ? The solution may be to create a special DialPlan but is there a way to modify the variable “DIALOPTIONS” depending of the extension that make the call ? Thanks, Fabien ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [November 16, 2010, 7:42pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/13 "2010-11-16T19:42:44Z")
I also have this problem with PIAF, running asterisk v 1.6x. Any idea of how to fix this using the FreePBX interface? ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [July 1, 2011, 7:36pm UTC](https://community.asterisk.org/t/pickup-detection-problem/32606/14 "2011-07-01T19:36:56Z")
I have this problem too, running PIAF, using multiple Nortel 1535 phones, calling a variety of numbers. I have tried a number of things, no of which seem to have any effect on the problem: - Upgraded phone firmware from 0.2.50 to 0.2.76 - Added “r” to dialout options, and verified via log that it is using this: [2011-07-01 12:11:32] VERBOSE[5743] pbx.c: – Executing [s@macro-dialout-trunk:19] Dial(“SIP/111-00000007”, “SIP/VP-SIPSJCA/+14155551212,300,r”) in new stack - Running Asterisk 1.8.2.3 - FreePBX 2.8.1 - Using VoicePulse Obviously this phone must work better for most people, or I would hear more complaints. Is it possible that this situation is specifically worse with VoicePulse?