Re: [Asterisk-Users] Little OT.. SER Question
And so the file said to the Brian... Let there be enlightenment: strip(5); That'll strip off the first 5... Characters... From the URI Joshua Colp On 11/10/05 1:25 PM, Brian C. Fertig [EMAIL PROTECTED] wrote: Anyone with SER knowledge could you point me in a direction to setup SER to rewrite the SIP URI? Currently I have the following [EMAIL PROTECTED] I am setting it so it does the change but its still showing up with the prefix. I need it to look like this: [EMAIL PROTECTED] I got xxx.xxx.xxx.xxx to change to yyy.yyy.yyy.yyy I just need the prefix to go away now.. ☺ ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can´t compile asterisk1.2beta2
This has already been discussed, you need to upgrade your GCC to 3 or higher. Joshua Colp From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rafael R. GV Sent: Friday, November 04, 2005 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Can´t compile asterisk1.2beta2 Hi ... .. . gcc -shared -Xlinker -x -o chan_modem_bestdata.so chan_modem_bestdata.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_agent.o chan_agent.c chan_agent.c: In function `__login_exec': chan_agent.c:1684: parse error before `char' chan_agent.c:1701: `agent_goodbye' undeclared (first use in this function) chan_agent.c:1701: (Each undeclared identifier is reported only once chan_agent.c:1701: for each function it appears in.) chan_agent.c:1708: `tmpoptions' undeclared (first use in this function) chan_agent.c:1714: `update_cdr' undeclared (first use in this function) chan_agent.c:1732: `context' undeclared (first use in this function) chan_agent.c:1737: `play_announcement' undeclared (first use in this function) chan_agent.c:1864: `filename' undeclared (first use in this function) make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/var/root/astbillFiles/asterisk/channels' make: *** [subdirs] Error 1 any idea??? thanks Rafael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_exosip2
Hello Harry, This is rather the wrong list to ask this... since this is Asterisk, not OpenPBX.org Chan_exosip2 though is something I'm basically designing to have 3 operating modes. Full server: Most closely resembles chan_sip in that it acts as a B2BUA Partial proxy: Extensions are mapped to SIP URIs and it acts as a proxy. Gateway: No authentication occurs (this is presumably done outside by a SIP proxy), incoming calls just get thrown into a context. Outgoing calls are down via SIP URI. Joshua Colp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: Tuesday, November 01, 2005 10:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] chan_exosip2 Hello, I read roadmap on www.openpbx.org. Does chan_exosip2 will be able to provide a real sip proxy ? What about asterisk solutions ? Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Montreal Meet Asterisk Get-Together
Hello Folks, I thought Id make a sorta announcement as Ill be in Montreal on a partial vacation/partial hangout/partial meet and greet thing. I thought it might be nice for all the people in the area, and perhaps those attending the Meet Asterisk thing to get together for supper and talk. If anyone is interested, respond to this post and well decide on a place/time/date etc. Meet the minds behind the sillyness on IRC! Muahahaha yeah Joshua Colp ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP connection
Hi Joseph, Here's a basic entry for you that you should be able to adapt. [mypeer] Type=peer Host=ip or hostname Context=where to send the call Disallow=all Allow=ulaw Insecure=very The insecure=very causes Asterisk to not do any authentication and trust it based on the IP. Joshua Colp On 10/16/05 1:22 PM, Joseph Rothstein [EMAIL PROTECTED] wrote: Geetings to all. I am having a hell of a time getting incoming SIP connections to work properly, and am hoping that someone can help me. Here is what I am using as a guide (from the wiki): Incoming SIP Connections When Asterisk receives an incoming SIP call, the SIP Channel Module first tries to find a [user] section matching the caller name (From: username), then tries to find a [peer] section matching the caller's IP address. If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf. I am mainly concerned with the second point. I want to match an incoming SIP connection to a particular IP address. I have tried just about everything, and the connection always goes to the default context, or the context defined at the top of the sip.conf file. I would like to be able to direct incoming SIP connection to a particular set of extensions. There is no username and password involved as there will be many users coming from this one IP. This is what I have tried recently: [sipin_test] type=peer defaultip=195.27.242.120 context=test_trunk deny=0.0.0.0/0.0.0.0 permit=195.27.242.120/255.255.255.255 dtmfmode=rfc2833 disallow=all allow=ulaw nat=no I have also tried changing what is inside the brackets to the IP address. I have tried many many different combinations of the above, but the IP address never seems to get picked up correctly. I am testing the SIP connection using sipsak. I realize that Asterisk is probably not the best SIP server to use, and plan on migration to SER, but if anyone can offer any suggestions I would really appreciate it. Regards to all, Joe ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Early Media in 100 Ringing
Hello Ronald, A 180 Ringing is something that should not have SDP because it's out of band signaling of the exact status of the call, ringing. The PSTN Gateway should return a 183 Session Progress if it wants to deliver inband audio progress. Their SIP implementation doesn't look the best either... so to get it to work you'd either have to hack Asterisk, or get the manufacturer of the PSTN gateway to fix their stuff. Joshua Colp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Voermans Sent: Monday, September 26, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Early Media in 100 Ringing Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? U 10.254.254.1:5060 - 192.168.0.173:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.0.173:5060;rport=5060;branch=z9hG4bK454e2d35. Record-Route: sip:[EMAIL PROTECTED]:5060. Record-Route: sip:[EMAIL PROTECTED]:5060;lr;nat=yes. From: 0161801019 sip:[EMAIL PROTECTED];tag=as02de1b95. To: sip:[EMAIL PROTECTED];tag=00-04094-52dbe3bc-6cf68a723. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Contact: sip:212.241.48.70:5060. server: Cirpack/v4.38f (gw_sip). Allow: UPDATE, REFER. Content-Type: application/sdp. Content-Length: 253. . v=0. o=cp10 112775383044 112775383045 IN IP4 10.166.38.109. s=SIP Call. c=IN IP4 10.254.254.1. t=0 0. m=audio 35058 RTP/AVP 18 101. b=AS:64. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. # U 192.168.0.173:5060 - 192.168.1.103:5062 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.103:5062;branch=z9hG4bKff31d98edbf2b265. From: 411 sip:[EMAIL PROTECTED];tag=f93ee2f65c6906cb. To: sip:[EMAIL PROTECTED];tag=as675f246d. Call-ID: [EMAIL PROTECTED] CSeq: 60590 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: sip:[EMAIL PROTECTED]. Content-Length: 0. . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Tandem Inbound only.
Hi Scott, To do what you want to do you do indeed need to use a peer entry, with the IP address where INVITEs will come from specified as the host, and insecure=very. Your OPTIONS though is being caused by qualify being turned on somewhere. Joshua Colp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Eisert Sent: Tuesday, September 27, 2005 4:48 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Tandem Inbound only. On Tuesday 27 September 2005 3:12 pm, Peter Bowyer wrote: On 27/09/05, Scott Eisert [EMAIL PROTECTED] wrote: Hello, I have a carrier that is supplying me with DID inbound over SIP to my asterisk server. Because the CID is different with every call that is coming in the only way I have to authenticate this carrier is IP based. In my sip.conf I want to define this user as type=user, however this can't work because Asterisk only authenticates users by username, not IP. Check out 'insecure=very' for sip.conf. Peter It doesn't look like insecure can solve my problem. If I have type=user, I send back a 404 regardless of the insecure setting. If I have type=peer or type=friend I can receive calls but asterisk sends out Options messages regardless of the insecure setting (yes or very). Any other suggestions? - Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Directed pickup syntax?
You have to tell it the extension you want to pick up, it's not psychic.
Doing what you're doing now would give the application no extension.
Exten = _*99.,1,Pickup(${EXTEN:3}) should work, with usage being
*99extension to pickup
Joshua Colp
On 9/24/05 7:28 PM, Rich Adamson [EMAIL PROTECTED] wrote:
What's the proper syntax for implementing directed call pickup?
Running cvs-head from today (9/24/05 including Mark's fixes), and
tried:
exten = *99,1,Pickup(${EXTEN:3})
but that does not seem to work, and there isn't an example in the
configs directory. 'show application pickup' suggests the above
should work with our sip phones, but apparently I'm missing
something.
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RE: [Asterisk-Users] T.38 Canreinvite (yes, again)
Hello, Asterisk does not act as a SIP Proxy as you may have in mind. Each call is treated independently, that is - codec capabilities of one call don't go to the other one during a reinvite. Only the IP address and Port go. Joshua Colp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, September 19, 2005 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] T.38 Canreinvite (yes, again) I know this has been asked before, but I've checked the archives and I haven't found anybody that has given a definitive yes or no, just yeah, it should work.. If I have a T.38 gateway like a Cisco 5300 and a T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work? I have it setup and it doesn't work, so I want to know if I am doing something wrong, or if it just won't work. If I make a voice call, I see the media stream go from the gateway to the ata directly. When I fax, I see the stream go that way as well, but it is g.729. I see INVITE messages from my ATA that reference T.38, but they go to the * box, not the gateway and therefore * ignores it. Any thoughts? PA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: R: [Asterisk-Users] direct sip call pickup
Hello Everyone, For regular call pickup you can't really specify a pickup group number... that's why it's set in the configuration. For directed call pickup you need to have the latest CVS head as it uses an API call that Kevin put in espically for me to use lastnight. Joshua Colp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Friday, September 16, 2005 2:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: R: [Asterisk-Users] direct sip call pickup What you were trying to do, *8#exten, is almost right I think. Look at it like this instead, though. The # is a pickup group number: *8x where x is the pickup group you want to pick up a call from. I could be wrong but that's how I understood it. Mojo Giordano Grandis wrote: I cannot use CVS, is there anoyher way to use direct pickup ? Thanks again **Giordano** *Da:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *Per conto di *Alexander Lopez *Inviato:* venerdì 16 settembre 2005 17.53 *A:* Asterisk Users Mailing List - Non-Commercial Discussion *Oggetto:* RE: [Asterisk-Users] direct sip call pickup On CVS head there is app_directed_pickup It will let you pickup a ringing extension directly without having to worry about pickup groups etc. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Giordano Grandis *Sent:* Friday, September 16, 2005 11:19 AM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] direct sip call pickup Hi, im working about sip call pick and *8 works very fine but I pickup ringing phone on the same group. What happen if I have more than one ringing call? I tryied *8#exten, *8eten# but it doesnt wotk. Is it correct? How it does work ? Thanks **Giordano** ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
