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jleed at me.com
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 Posted: Tue Feb 24, 2015 8:53 am Post subject: [asterisk-users] having trouble to register cisco 7975 with |
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Ok after I added tcp transport and disable force_rport phone get registered, but still have issues with calls,
when I call from cisco from, it work except hangup.
when I call to cisco phone asterisk return congested
debug of call
<--- Transmitting SIP request (952 bytes) to TCP:192.168.1.61:51179 --->
INVITE [url=sip:111@192.168.1.61:51179;transport=tcp]sip:111@192.168.1.61:51179;transport=tcp[/url] SIP/2.0
Via: SIP/2.0/TCP 192.168.1.4:55246;rport;branch=z9hG4bKPjcb9ec9ba-0c77-4530-a3b7-44209357f3a0;alias
From: <[url=sip:502@192.168.1.4]sip:502@192.168.1.4[/url]>;tag=abebd75c-501a-4b4f-ad69-ee98175b8dbd
To: <[url=sip:111@192.168.1.61]sip:111@192.168.1.61[/url]>
Contact: <[url=sip:28552048-b20b-4e7c-8454-f7d1486fd8ef@192.168.1.4:55246;transport=TCP]sip:28552048-b20b-4e7c-8454-f7d1486fd8ef@192.168.1.4:55246;transport=TCP[/url]>
Call-ID: bb515935-7292-47b4-890d-6f82eb335815
CSeq: 25333 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 283
v=0
o=- 1231372975 1231372975 IN IP4 192.168.1.4
s=Asterisk
c=IN IP4 192.168.1.4
t=0 0
m=audio 17856 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
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=maxptime:150
a=sendrecv
[Feb 24 05:47:01] WARNING[16179]: pjsip:0 <?>: tsx0x7f1aa0157 Failed to send Request msg INVITE/cseq=12216 (tdta0x7f1aa00e41c0)! err=120111 (Connection refused)
[Feb 24 05:47:01] ERROR[16179]: pjsip:0 <?>: tcpc0x7f1aa01c TCP connect() error: Connection refused [code=120111]
[Feb 24 05:47:01] WARNING[16179]: pjsip:0 <?>: tsx0x7f1aa01c3 Failed to send Request msg INVITE/cseq=25333 (tdta0x7f1aa00ad810)! err=120111 (Connection refused)
| Quote: | On 24 Feb 2015, at 15:05, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
Nick Awesome wrote: | Quote: | | Hay guys, got trouble with registration with cisco 7975 |
The "force_rport" option is incompatible with Cisco, it needs to be explicitly set to no in the endpoint.Cheers,-- Joshua ColpDigium, Inc. | Senior Software Developer445 Jan Davis Drive NW - Huntsville, AL 35806 - USCheck us out at: www.digium.com & www.asterisk.org-- _____________________________________________________________________-- 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/helloasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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jcolp at digium.com
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| Posted: Tue Feb 24, 2015 9:03 am Post subject: [asterisk-users] having trouble to register cisco 7975 with |
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Nick Awesome wrote:
| Quote: | Ok after I added tcp transport and disable force_rport phone get
registered, but still have issues with calls,
when I call from cisco from, it work except hangup.
when I call to cisco phone asterisk return congested
|
If you use UDP with force_rport=no it'll work.
If you use TCP then set rewrite_contact=yes so it'll reuse the
established TCP connection.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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