search for: 5070

Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/25402ade/attachment.htm

I am playing with tcpenable... on 13.11.2 so in sip.conf I have tcpenable=yes tcpbindaddr=192.168.1.8:5070 but when I "telnet localhost 5070" I get no connect. iptables -L -n -v | grep 5070 0 0 ACCEPT tcp -- * * 0.0.0.0/0 0.0.0.0/0 state NEW tcp dpt:5070 firewall is good. Is my syntax not correct above to run on port 5070 for SIP over TCP? Thanks, Jerr...

...[2020-09-19 23:42:19] VERBOSE[2637] chan_sip.c: <--- SIP read from UDP:41.11.11.12:5060 ---> INVITE sip:0100000000 at 52.22.22.22:5160 SIP/2.0 Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK4df7.5bc77035.0 Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070 From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614 To: <sip:0100000000 at 52.22.22.22:5160> Contact: <sip:+27888888888 at 41.11.11.11:5070> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.1...

Hi All, I have a process running on port 5070... I'm using CentOS 7. iptables is running firewalld should be stopped and disabled. When I telnet localhost 5070 I get connection refused. When I stop iptables I still get connection refused. netstat -tnlv | grep 5070 tcp 0 0 192.168.1.8:5070 0.0.0.0:* LISTE...

...? The following is the * CLI> log to 192.168.0.161:43222 -- Executing NoOp("SIP/4881-bde9", "") in new stack -- Executing RxFAX("SIP/4881-bde9", "/home/zenkato/voip/asterisk/fax/tif/send22.tif") in new stack Sip read: ACK sip:4883@192.168.0.3:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161:43222;branch=z9hG4bK3d3700001a29ffff From: <sip:4881@192.168.0.3:5070>;tag=bdf000008f360000 To: <sip:4883@192.168.0.3:5070>;tag=as4090e42f Contact: <sip:4881@192.168.0.161:43222> Proxy-Authorization: DIGEST username="4881", realm=&q...

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen and tried before sometimes.sip-happens.com and additionally, when the connect...

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it 151@myIP:5070, and asterisk will match it to 151 in the dialplan. In asterisk 1.2 asterisk completely ignores the request (even at mos...

...nternal calls (which would be preferable in this case) or after the 488 sent by Asterisk I'd need Asterisk to relay the sdp offered by Kamailio/rtpengine as such without rewriting it. Here the call works fine (internal call from sip peer 771 to webrtc peer 660): INVITE that Asterisk (at port 5070) receives: PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 INVITE sip:660 at testers.com;transport=UDP SIP/2.0 Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0 Via: SIP/2.0/UDP AST.ER.ISK.IP:38699;r...

...a Re-Invite for the same call. I respond to the 491 response with an ACK however for some strange reason Asterisk doesn't accept the ACK and insists on retransmitting the 491 Response. Asterisk replies with the following 491 response: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 10.110.7.89:5070;branch=z9hG4bK5a668c33f196837c3602266b23b389e0;received=10.110.7.89 From: <sip:30001 at 10.110.7.20:5070>;tag=SIPTester To: <sip:30008 at 10.110.7.20>;tag=as2ea72122 Call-ID: 0dd43bb5a64eb5a2fb0114193821f037 at 10.110.7.89 CSeq: 5 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK...

...s. What is the reason for such response? What are the criteria for such evaluation? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP 192.168.129.74:5070;branch=z9hG4bK5b33ae78;rport=5070 From: "IPFon" <sip:0225761853 at 192.168.129.74:5070>;tag=as7217acbc To: <sip:tzl at voip.touk.pl>;tag=as7217acbc Call-ID: 307fda656066a7e264e85cea0742e601 at 192.168.129.74 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL,...

...the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0 To: <sip:0041787518551 at 192.168.0.110> From: 021111111 <sip:021111111 at peoplefone.ch >;tag=27B98752-469CEA8A0002F2E4-5F903B30 CSeq: 10 INVITE Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30 at 212.203.123.82 Content-Length: 250 User-Agent: OpenSER (1.2.1-no...

...[2020-09-19 23:42:19] VERBOSE[2637] chan_sip.c: <--- SIP read from UDP:41.11.11.12:5060 ---> INVITE sip:0100000000 at 52.22.22.22:5160 SIP/2.0 Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK4df7.5bc77035.0 Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070 From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614 To: <sip:0100000000 at 52.22.22.22:5160> Contact: <sip:+27888888888 at 41.11.11.11:5070> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.1...

...routes) and I think I found a bug in the way Asterisk prepare new requests inside a dialog. I'm sending some captures (ngrep) along with my comments. This is a 200 OK (INVITE) received by Asterisk ========================= U 2005/02/10 16:41:55.065538 143.173.202.82:5060 -> 143.173.202.83:5070 SIP/2.0 200 OK..Via: SIP/2.0/UDP 143.173.202.83:5070;branch=z9hG4bK42c78895..Record-Route: <sip:001178612341106@143.173. 202.81;ftag=as182aa61c;lr>..Record-Route: <sip:001178612341106@143.173.202.82:5060>..From: "Call Center 1" <sip:55512@si p.trdc.telenova.com.br>...

I made a similar question in a previous thread, but there was no answer, so I think I was not very clear making the question. What I need is some configuration that works like "promiscredir=yes" in sip.conf that enables me to do the same thing with transfer (REFER), letting me transfer a sip call to a non local sip address. Thanks in advance, Thiago Abra sua conta no Yahoo!

Apparently Verizon is blocking or changing packets on port 5060 so my softphone from my hotspot will not work. How do I set asterisk (11.23.0) to run default 5060 for all other devices I have - BUT for my software run on a different port like 5070? I'm using linphone and is easy to change the ports from 5060 to 5070 ( I think). Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161014/116521bc/attachment.html>

...[2020-09-19 23:42:19] VERBOSE[2637] chan_sip.c: <--- SIP read from UDP:41.11.11.12:5060 ---> INVITE sip:0100000000 at 52.22.22.22:5160 SIP/2.0 Record-Route: <sip:41.11.11.12;lr;ftag=as40fe2614> Via: SIP/2.0/UDP 41.11.11.12:5060;branch=z9hG4bK4df7.5bc77035.0 Via: SIP/2.0/UDP 41.11.11.11:5070;received=41.11.11.11;branch=z9hG4bK0cb77ea3;rport=5070 From: "+27888888888" <sip:+27888888888 at 41.11.11.11:5070>;tag=as40fe2614 To: <sip:0100000000 at 52.22.22.22:5160> Contact: <sip:+27888888888 at 41.11.11.11:5070> Call-ID: 7030be5a09d89a9543234da051897a49 at 41.11.1...

...ow can we establish two way audio? - After this re-invite Asterisk replies with a "100 Trying" and then a "200 OK" which contains "a=recvonly". - Call is established but called party cannot hear caller. Here's the re-invite message - note that Asterisk is on port 5070: U 2010/05/05 12:47:38.139701 (peer):5060 -> (asterisk):5070 INVITE sip:(called number)@(asterisk):5070 SIP/2.0. Via: SIP/2.0/UDP (peer):5060;branch=z9hG4bK2sansay7330954rdb6594. To: "User" <sip:(called number)@(asterisk):5070>;tag=as3ddcc528. From: <sip:(called number)@(p...

...erisk box to accept arbitrary sip URI in a REFER (xfer) command. Right now it always tries to send the call to a local extension, for example, if I have a call from my asterisk to "555 at 10.10.10.1:5060" and this application asks asterisk to transfer this call to "666 at 10.10.10.2:5070" asterisk will try to send the to the local extension 666. Bellow I have a sip debug from the messages. My asterisk box is running in the IP 201.73.67.5, and my first application (the one that asterisk dials directly) is at the address 201.73.67.7:5080 and it transfers the calls to 201.73.67.7...

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to