search for: 5062

...is sent in response to a 200 message. The full SIP dialog is at http://pastie.org/private/nybdytnfyfenovpwfywcya so as to not clutter the email, but I have included the highlights below: >>> The call was ringing and is now answered: <--- Reliably Transmitting (NAT) to 82.158.83.xxx:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 82.158.83.xxx:5062;branch=z9hG4bK1226703311;received=82.158.83.xxx;rport=5062 From: "800902" <sip:800902 at 130.117.xxx.xxx;user=phone>;tag=467506068 To: <sip:6615xxxxx at 130.117.xxx.xxx;user=phone>;tag=as2e12c791 Call-ID: 2117388659-50...

Hi Friends, I want to use 5062 port for SIP protocol. I made the below modifications in my server to use 5062 port. Polycom phone: port=5062 Trunk settings: port=5062 sip.conf: bindaddr=5062 Extension configuration details: 5062 Our VoIP provider told me that they are allowing the SIP traffic through 5060 to 5064. I observe...

...t;<"David" <sip:>> . ERROR:parse_from_header: bad from header ERROR: new_t: no valid From in INVITE ERROR: t_newtran: new_t failed ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) I got nex sip messages: U 2006/01/16 12:21:10.968713 10.2.11.35:5062 -> 10.2.11.35:5060 INVITE sip:204@10.2.11.35 SIP/2.0..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK31f811a3;rport..From: "Zyxel" <sip:Zyxel<sip:204@1021150;user=phone@10.2.11.35:5062>;tag=as56432543..To: <sip:204@10.2.11.35>..Contact: <sip:Zyxel<sip:204@1 02...

...o reproduce this while > sip debugging was on, so I have that information available now as well: > http://pastebin.com/ZJqzdvY3 > > This was a call from 113 to 146 via a queue. Note that the asterisk server is > at 10.10.32.251. I see the following: > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > SIP/2.0 180 Ringing > SIP/2.0 180 Ringing > SIP/2.0 200 OK > ACK sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > SIP/2.0 200 OK > ACK sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip...

Hi all! Am I missing some configuration or is it simply a bug: If Asterisk chan_sip is configured with bindport=5062, the port is missing on the outgoing SIP messages contact header. This resulting in in-dialog messages sent to port 5060 ... where there is no Asterisk on that host... Tried externip = 1.2.3.4:5062 with no success. Version 1.6.0.3. br Walter -------------- next part -------------- An HTML a...

Hello, I am running Asterisk 11 on CentOS 6.4 with SIP clients (Yealink phones). All the SIP clients are on a LAN, so no NAT is involved. I have been experiencing an intermittent problem where a call will be successfully answered, but then dropped by Asterisk 32 seconds after it is answered (with a "Retransmission timeout reached on transmission" error). Here is an example of this

...################################################################# ###################### RESTRICTED SIP ACCESS ################################ ############################################################################# # LAN /sbin/iptables -A INPUT -p tcp -i eth0 -s 192.168.1.0/24 --dport 5060:5062 -j ACCEPT /sbin/iptables -A INPUT -p udp -i eth0 -s 192.168.1.0/24 --dport 5060:5062 -j ACCEPT /sbin/iptables -A INPUT -p udp -i eth0 -s 192.168.1.0/24 --dport 10000:20000 -j ACCEPT # Allow traffic from VoIP Service Provider /sbin/iptables -A INPUT -p udp -i eth1 -s 11.11.11.11 --dport 5060:...

...and AMP to manage it. I've made 4 extensions: moloch*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN 203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms) 202/202 (Unspecified) D 255.255.255.255 0 UNKNOWN 201/201 192.167.125.12 D 255.255.255.255 5060 OK (3 ms) moloch*CLI> as you can see, 201 and 203 are on-line but, if i call from 203 to 201, i immediately go to voicemail...

...y Host Username Refresh State moloch*CLI> If i try, from 203, calling 201 this is what happens: ===========================8<=================================== moloch*CLI> Sip read: INVITE sip:201@asb.unisi.it SIP/2.0 Via: SIP/2.0/UDP 192.167.125.9:5062;branch=z9hG4bK78ED0E83 CSeq: 7665 INVITE To: <sip:201@asb.unisi.it> Content-Type: application/sdp From: "203" <sip:203@asb.unisi.it>;tag=2B558754 Call-ID: 499575437@192.167.125.9 Subject: sip:203@asb.unisi.it Content-Length: 187 User-Agent: kphone/4.0.5 Contact: "203&quot...

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

Greetings I have been running * for about a month now. Configuration. (5) Cisco 79xx IP phones (1) XP100P Pentium III (300mhz) 192meg memory Redat 8.0 (updated) It seems to run for about 3-6 hours, then the process stops. I have noticed, that * does not stop, if I do NOT have it register to other sip servers. (FWD and PCH). Here is are the last few lines in the /var/log/asterisk/messages

..., SUBSCRIBE, NOTIFY Contact: <sip:699@192.168.1.10> Content-Length: 0 --- -- Executing Dial("SIP/677-bd65", "sip/699") in new stack We're at 192.168.1.10 port 12042 Adding codec 0x4 (ulaw) to SDP 13 headers, 8 lines Reliably Transmitting (no NAT) to 192.168.1.201:5062: INVITE sip:699@192.168.1.201:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK036b03c0;rport From: "John Millican" <sip:677@pap2>;tag=as604497e3 To: <sip:699@192.168.1.201:5062> Contact: <sip:677@192.168.1.10> Call-ID: 400784d110e4e314373334be31ee3576@pap2...

Hi Mark, Current CVS, * isn't correctly remembering the tag added to the To header by a server. For instance: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09 From: steve-ata186 <sip:asterisk@81.96.69.210:5062>;tag=14925711 To: <sip:18478974611@4.42.235.170>;tag=t2907cab0911c8g Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210 CSeq: 102 INVITE Allow: INVITE, CANCEL, REFER, BYE, ACK Contact: <sip:0403242876@195.217.255....

Hello, Hope somebody can help me - Asterisk is behaving very oddly and I'm totally stumped! I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a different port. It is my understanding that I just need to set the port in sip.conf (port=5062) but that doesn't seem t...

...rrier no problems, however, receiving calls doesn't work, and I keep getting the following messages: <-- SIP read from 69.xx.xx.xx:5060: INVITE sip:s@10.0.0.200:5060;maddr=10.0.0.200 SIP/2.0 Record-Route: <sip:83555501@69.xx.xx.xx:5060;maddr=69.xx.xx.xx>, <sip:83555501@69.xx.xx.xx:5062;maddr=69.xx.xx.xx> Via: SIP/2.0/UDP 69.xx.xx.xx:5060, SIP/2.0/UDP 69.xx.xx.xx:5062, SIP/2.0/UDP 69.xx.xx.xx:5062, SIP/2.0/UDP 69.xx.xx.xx:5081;branch=z9hG4bKB18439727BD34C17BE5095ACF45BE007 To: <sip:83555501@69.xx.xx.xx:5060> From: Sason <sip:grouphone0@69.xx.xx.xx:5081>;tag=KmsqOjY5...

...ugging was on, so I have that information available now as well: > > http://pastebin.com/ZJqzdvY3 > > > > This was a call from 113 to 146 via a queue. Note that the asterisk server > > is > > at 10.10.32.251. I see the following: > > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > > SIP/2.0 180 Ringing > > SIP/2.0 180 Ringing > > SIP/2.0 200 OK > > ACK sip:146 at 10.10.32.96:5062 SIP/2.0 > > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > > SIP/2.0 200 OK > > ACK sip:146 at 10.10.32.96:5062 SIP/2.0 > > INVITE sip:146 at...

...voice aors = gv-voice identify_by=auth_username ;identify_by=username ; I also tried this. Same result. context = gv-voice [gv-voice](obi202-auth) username = gv-voice [gv-voice](obi202-aor) ############## From the pjsip logging: <--- Received SIP request (798 bytes) from UDP:<obi_ip>:5062 ---> INVITE sip:<gv_num>@<ast_ip>:5060 SIP/2.0 Call-ID: bb384ee02eab7054 at 10.10.11.181 Content-Length: 270 CSeq: 8001 INVITE From: <sip:+1<calling_num>@<ast_ip>>;tag=SP377bfeeed75f36b8e Max-Forwards: 70 To: <sip:<calling_num>@<ast_ip&gt...

...calls today, I have managed to reproduce this while sip debugging was on, so I have that information available now as well: http://pastebin.com/ZJqzdvY3 This was a call from 113 to 146 via a queue. Note that the asterisk server is at 10.10.32.251. I see the following: INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 SIP/2.0 180 Ringing SIP/2.0 180 Ringing SIP/2.0 200 OK ACK sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 SIP/2.0 200 OK ACK sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 INVITE sip:1...

...after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1: Operation not permitted [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b704e0 (len 543) to my_ip:5062 returned -1: Operation not permitted [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b722b0 (len 543) to my_ip:5061 returned -1: Operation not permitted [Dec 23 11:11:07] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of 0x7b6e710 (len 549) to my_ip:5063 returned -1...

...ed asterisk 1.6 to 1.8. As the result of this when peers trying to register to asterisk the system shows: NOTICE[24698]: chan_sip.c:23417 handle_request_register: Registration from '"50" <sip:50 at 192.168.1.109> <sip:50 at 192.168.1.109>' failed for ' 192.168.1.80:5062' - Wrong password even though on 1.6 everything was OK here is part of debug messages: ---cut--- <--- Transmitting (NAT) to 192.168.1.50:5062 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.50:5062;branch=z9hG4bK-d8754z-d7545057f425cd49-1---d8754z-;received=192.168.1.50;rport=50...