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...begins) Sip read: REGISTER sip:192.168.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0 Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (4asipfxs.107a) Contact: <sip:1234@192.168.0.202:5060>;expires=60 From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48a3 To: <sip:1234@192.168.0.200> Call-ID: c0a800ca-13c4-0-334-1c34 CSeq: 1 REGISTER Content-Length:0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.202 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202:5060;branc...

...:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:1649;phone-context=Exp_Net.ascom@ascom.be;user=phone>;tag=de6719ac-13c4-45630b31-5f87fa39-180d To: <sip:1715;phone-context=Exp_Net.ascom@ascom.be;user=phone> Call-ID: f8b5ad8-de6719ac-13c4-45630b31-5f87fa39-d37@ascom.be CSeq: 1 INVITE Via: SIP/2.0/UDP 172.25.103.222:5060;branch=z9hG4bK-45630b31-5f87fa39-63c2 Max-Forwards: 70 Supported: 100rel,sipvc,replaces,timer...

...ce! michelle PD:188.208.12.237 is the asterisk IP (...) s=session c=IN IP4 188.208.12.237 t=0 0 =audio 13532 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 229.159.241.112:5060 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-3a5246f7- 8c6b606-10eb From: ;tag=0-13c4-3a5246f7-8c6b604-c3a To: ;tag=as52ed0a6a Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765@188.208.12.37 CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: Content-Type: application/sdp Content-Length: 135 v=0 o=root 11673 11673 IN IP4 188.208.12.237 s=session c=IN IP4 188.208.12.237 t=0 0 =audio 13532...

...k 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To: <sip:2828;phone-context=cdp.udp@qg.com;user=phone> Call-ID: 103548e8-c22da8c0-13c4-429efa71-657b8da-517@qg.com CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.45.194:5060;branch=z9hG4bK-429efa71-657b8da-2522 Max-Forwards: 70 Supported: 100rel,sipvc,replaces User-Agent: Nort...

...lify=yes ------------------------------------------------------------------- Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To: <sip:2828;phone-context=cdp.udp@qg.com;user=phone> Call-ID: 103548e8-c22da8c0-13c4-429efa71-657b8da-517@qg.com CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.45.194:5060;branch=z9hG4bK-429efa71-657b8da-2522 Max-Forwards: 70 Supported: 100rel,sipvc,replaces User-Agent: Nortel...

..., CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- ? [Khfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 ---> SIP/2.0 200 OK From: "asterisk"<sip:asterisk@161.49.142.250>;tag=as2cc96e52 To: <sip:192.168.45.129>;tag=812da8c0-13c4-46277c06-279cd106-42ff Call-ID: 3ee92dbe77f51a1748f736be4593719d@161.49.142.250 CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK2508d83c Supported: 100rel,sipvc,replaces User-Agent...

...is numbers - the call goes, but when I pickup phone - communication breaks... Please, help to understand! Here a log: <-- SIP read from 172.222.135.11:5060: SIP/2.0 200 OK From: "Unknown"<sip:Unknown@10.11.0.150>;tag=as2335e618 To: <sip:3301@172.222.135.11>;tag=ac16230b-13c4-ad-2c457-4966 Call-ID: 08001b0456b9377a479050064635a26e@10.11.0.150 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.11.0.150:5060;rport=5060;branch=z9hG4bK0e38ec58 Supported: replaces User-Agent: FXS_GW (2sipfxs.112) Contact: <sip:3301@172.222.135.11:0> Content-Type: application/sdp Content-Length: 220...

...he second media stream or is there some more configuration tweaking to be done? --- CUT ---------------------------------------------- <--- SIP read from UDP:192.168.1.13:5060 ---> INVITE sip:1XXXXXXXXXX at 192.168.1.11 SIP/2.0 From: Biscom <sip:418 at 192.168.1.13>;tag=86c9140-d281eac-13c4-55013-1f571-33180d4a-1f571 To: <sip:1XXXXXXXXXX at 192.168.1.11> Call-ID: 73cb2e8-d281eac-13c4-55013-1f571-7ef285e4-1f571 CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-1f571-7a6c245-2c788e59 Supported: 100rel Max-Forwards: 70 User-Agent: Brktsip/6.6.9B12 (Dialogic) Contact:...

...ation from '<sip:101@192.168.100.240>' failed for '192.168.100.104' - Username/auth name mismatch Sip debug show the following: <-- SIP read from 192.168.100.104:5060: REGISTER sip:192.168.100.240:5060 SIP/2.0 From: <sip:101@192.168.100.240>;tag=c0a86468-13c4-43f1a32d-c2e70-35fb To: <sip:101@192.168.100.240> Call-ID: c0a86468-13c4-c-334f-370 CSeq: 14 REGISTER Via: SIP/2.0/UDP 192.168.100.104:5060;branch=z9hG4bK-43f1a32d-c2e75-2618 Max-Forwards: 70 Supported: replaces User-Agent: IP_PHONE (vip150t_050413.bin) Contact: <sip:101@192.168.10...

...a recalculation when I quit Thunderbirdb and I relaunch it. Even, with doveadm CLI, as long as Thunderbird is not disconnected on the client side, the server didn't recalculate the storage available. Thanks for the help ! Thomas -- Thomas Criscione https://thomascriscione.fr GPG: E521 FFF0 13C4 52FB 55EB 8C49 EBDD 0DC6 1ED9 2A4F

...P]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden (From header is not a Trust host or gateway) From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]>;tag=87f2a0d5-13c4-4146e24d-1a7b91cc-796f Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines -- Got SIP response 403 "Forbidden (From header is not a Trust host or gateway)" back from [...

...INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Date: Wed, 23 Apr 2014 13:20:59 GMT Contact: <sip:3145152000 at 192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68> Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4 Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967 Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947 Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4b...

...d. However, if another machine on the LAN opens a client no client will work. Attempting to call anything like Voicemail fails and after a short while Asterisk starts scrolling: [Feb 10 11:10:31] WARNING[8852]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1064dc5c-5101a8c0-13c4-3ba4-e88578-5c0 at 192.168.1.81 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [Feb 10 11:10:31] WARNING[8852]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1064dc5c-5101a8c0-13c4-3ba4-e88578-5c0 at 192.168.1.81 for seqno 1 (Critical Response) -- See doc/sip...

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have

...102 INVITE Via:SIP/2.0/UDP [my IP]:5060;branch=z9hG4bK00215868 Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden (From header is not a Trust host or gateway) From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:69101701@[voip IP]>;tag=87f2a0d5-13c4-4146d5ea-1a4b30eb-3af4 Call-ID: 0013b54e26506f7f4133cc0f59f1e561@[my IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my IP]:5060;branch=z9hG4bK00215868 Content-Length:0 ( [my IP] is my external IP, [voip IP] is the IP of the SIP server of my VoIP provider. [dialled number] is the number I dialled) I don&...

...all kinds of combinations of user,username,authname using +15552027020,host with IP and/or DNS name, but nothing appears to work. Frank INVITE message from Wireshark packet capture: INVITE sip:+15552027020 at sip.acme.com SIP/2.0 From: <sip:5552022441 at 172.16.10.40>;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db ba4 To: <sip:+15552027020 at sip.acme.com> Call-ID: f379f62-29173-3895-b14271f5-40802-45378 at 172.16.10.40 CSeq: 5102 INVITE Via: SIP/2.0/UDP 172.16.10.40:5060;branch=z9hG4bK-2dbba4-b2a4fa3a-7cd7598 User-Agent: Nortel CS1500UA/v02.00.REL01 Accept: application/sdp P-Assert...

...opped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum retries exceeded on call 2002fb00-4d0b478-13c4@leadtek.com.tw for seqno 102 (Non-critical Request)

...identical behavior with callgroup=/pickupgroup= specified for each extension. Here's some sanitized verbose output with SIP debugging enabled: -- Starting simple switch on 'Zap/24-1' Jun 28 10:43:18 DEBUG[16774]: chan_sip.c:771 __sip_autodestruct: Auto destroying call 'a01052a-13c4-42c104ea-371e-1957' Destroying call 'a01052a-13c4-42c104ea-371e-1957' Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/24-1 Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit: 3 on Zap/24-1 Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read:...

...> > Even, with doveadm CLI, as long as Thunderbird is not disconnected on > the client side, the server didn't recalculate the storage available. > > Thanks for the help ! > > Thomas > > -- > Thomas Criscione > https://thomascriscione.fr > GPG: E521 FFF0 13C4 52FB 55EB 8C49 EBDD 0DC6 1ED9 2A4F