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Call Transfer Does Not Work

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Hi,

 

i am having a very interesting issue but this is an urgent one.

 

The client is running Windows System Version 4.2.0.3961 (Win32) and using all Polycom Phones.

 

The clients receives a call and can talk regularly with the the caller however if they attempt to transfer the caller hears hold music and ringback until the second party picks up the phone but once the second user picks up the phone there is dead air on both ends!

 

please see extract of the logfile below.

 

SIP/2.0 200 OK

Via: SIP/2.0/TCP 10.0.3.16:5060;branch=z9hG4bK-df21e7b7aba9f122f5119b08b53f9e9d;rport

From: "Susan Currie" <sip:325@localhost>;tag=16259

To: "Doreen Agliata" <sip:319@localhost>;tag=7C41E843-809506C6

CSeq: 28452 INVITE

Call-ID: c52aa411@pbx

Contact: <sip:319@10.0.3.51;transport=tcp>

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

Supported: 100rel,replaces

User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.2.1.0054

Accept-Language: en-us,en;q=0.9

Content-Type: application/sdp

Content-Length: 193

 

v=0

o=- 1292261604 1292261604 IN IP4 10.0.3.51

s=Polycom IP Phone

c=IN IP4 10.0.3.51

t=0 0

m=audio 2226 RTP/AVP 0 101

a=sendrecv

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

[7] 2010/12/13 12:40:13: Call c52aa411@pbx: Clear last INVITE

[7] 2010/12/13 12:40:13: SIP Tx tcp:10.0.3.51:51811:

ACK sip:319@10.0.3.51;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 10.0.3.16:5060;branch=z9hG4bK-d174e72aa06f2b9b1b1f8aa27b7cb9d2;rport

From: <sip:7328008008@localhost:5060;transport=udp;user=phone>;tag=16259

To: "Doreen Agliata" <sip:319@localhost>;tag=7C41E843-809506C6

Call-ID: c52aa411@pbx

CSeq: 28452 ACK

Max-Forwards: 70

Contact: <sip:319@10.0.3.16:5060;transport=tcp>

Content-Length: 0

 

[8] 2010/12/13 12:40:14: DNS: Add NAPTR callcentric.com (ttl=60)

[8] 2010/12/13 12:40:14: DNS: Add SRV _sips._tcp.callcentric.com (ttl=60)

[8] 2010/12/13 12:40:14: DNS: Add SRV _sip._tcp.callcentric.com (ttl=60)

[7] 2010/12/13 12:40:19: SIP Rx udp:10.5.4.50:5060:

OPTIONS sip:metaswitch@10.128.1.202:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.5.4.50:5060;rport;branch=z9hG4bK-d6deca0e45ed19f515a24c1897b171d7-meta2-o.pins.net-1

Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event

Max-Forwards: 70

Call-ID: BB5EF132@meta2-o.pins.net

From: <sip:metaswitch@10.5.4.50:5060;transport=udp>;tag=meta2-o.pins.net+1+0+9a600b28

CSeq: 575142591 OPTIONS

Organization:

Supported: 100rel, resource-priority

Content-Length: 0

Contact: <sip:metaswitch@10.5.4.50:5060;transport=udp>

To: <sip:metaswitch@10.128.1.202>

 

[7] 2010/12/13 12:40:19: SIP Tx udp:10.5.4.50:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 10.5.4.50:5060;rport=5060;branch=z9hG4bK-d6deca0e45ed19f515a24c1897b171d7-meta2-o.pins.net-1

From: <sip:metaswitch@10.5.4.50:5060;transport=udp>;tag=meta2-o.pins.net+1+0+9a600b28

To: <sip:metaswitch@10.128.1.202>;tag=14cc23a9cc

Call-ID: BB5EF132@meta2-o.pins.net

CSeq: 575142591 OPTIONS

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

Content-Length: 0

 

[5] 2010/12/13 12:40:27: Did not receive ACK, disconnecting call 46506489-6cfdc1d4-e75f677b@10.0.3.45

[8] 2010/12/13 12:40:27: Hangup: Call 224 not found

[7] 2010/12/13 12:40:35: SIP Rx tcp:10.0.3.34:57885:

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as a work around:

snom ONE built in star code will transfer call i understand.

put call on hold and dial the * code for transfer.

 

the whole idea is that non snom devices that are allowed are meant for things like doorphone, cameras, etc.

 

I'm not saying I like this but is how it currently works.

If you have a suggestion here: http://snomone.ideascale.com/

 

Matt

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as a work around:

snom ONE built in star code will transfer call i understand.

put call on hold and dial the * code for transfer.

 

the whole idea is that non snom devices that are allowed are meant for things like doorphone, cameras, etc.

 

I'm not saying I like this but is how it currently works.

If you have a suggestion here: http://snomone.ideascale.com/

 

Matt

 

 

 

Hi

 

I Just tried it on pnxnsip version 4.2.0.3961 (Win32) and i had the same bug on the polycom phone

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Have not heard back?

We were not able to reproduce it in a slightly later version (.3966). But there seems to be no change in that area. Only difference is that one of the phone used is snom & the other one is Polycom 331 (just had only 1 Poly)

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We were not able to reproduce it in a slightly later version (.3966). But there seems to be no change in that area. Only difference is that one of the phone used is snom & the other one is Polycom 331 (just had only 1 Poly)

 

please post a link for (.3966). for windows 32 and Linux (CentOS 5 32)

 

 

Thank You

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please post a link for (.3966). for windows 32 and Linux (CentOS 5 32)

 

 

Thank You

 

Not sure if you wanted snomone or pbxnsip versions. But here they are anyways -

 

snomone

Win32 – http://pbxnsip.com/snomone/beta/win32/pbxctrl-2011-4.2.0.3966.exe

CentOS32 - http://pbxnsip.com/snomone/beta/centos32/pbxctrl-centos5-2011-4.2.0.3966

 

pbxnsip

Win32 – http://pbxnsip.com/protect/beta/win32/pbxctrl-4.2.0.3966.exe

CentOS32 - http://pbxnsip.com/protect/beta/centos32/pbxctrl-centos5-4.2.0.3966

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I upgraded to 4.2.0.3966 (Win32) and below in the problems that i have

Call Transfer ( Polycom)

when trying to transfer a call from ext 2088 to ext 2089 not blind and when ringing i press Transfer again before the other side will receive the call when no audio (this problem is only if you transfer between 2 polycom phones

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Is this a transfer with early media (the call not connected yet)? This is tricky in SIP. If you can get us a PCAP or a LOG with the SIP messages (attach it, please) we can find out what the problem is.

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Is this a transfer with early media (the call not connected yet)? This is tricky in SIP. If you can get us a PCAP or a LOG with the SIP messages (attach it, please) we can find out what the problem is.

 

I sent you a email with the trace

 

please advise

 

Thank You

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Well, thats a SIP problem. How do you successfully disconnect a call that is not connected yet? We could send 487, right now we are sending 486. Not sure if this would make a big difference.

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Well, thats a SIP problem. How do you successfully disconnect a call that is not connected yet? We could send 487, right now we are sending 486. Not sure if this would make a big difference.

 

I checked 4.1.0.4026 (Win32) and you are send 486 and i looks like its the best we can do with the polycom for now

 

 

 

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