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...wav, 0x8144980 Mar 6 18:41:45 WARNING[3539]: app.c:619 ast_play_and_record: No audio available on SIP/69.70.x.x-08149a98?? -- User hung up == Spawn extension (ser, 900, 1) exited non-zero on 'SIP/69.70.x.x-08149a98' Destroying call 'ixiXpRvNGSyIBxmn@192.168.1.103' If I use rewritehostport instead of forward, the call does not reach asterisk: failure_route[1] { revert_uri(); rewritehostport("69.70.x.x:5060"); t_relay() break(); SER log: 4(11513) ******* IP to IP call ************* 1(11506) ERROR: t_forward_nonack: no branched for fwding...

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW: SER Asterisk Voicemail Date: Thu, 10 F...

...g to bundle the powers of Asterisk and SER. Asterisk for pabx functionalities and termination to landline/PSTN, and SER as SIP Gateway/Proxy. With my current configuration the SIP user just adds 0 as a prefix to a number, and the call will go out to PSTN over Asterisk. For this to work I added the rewritehostport() function in SER to point to the Asterisk IP (different from the SER ip). At the moment I just added the following line to my sip.conf (in the [general] section): context=from-sip But my question here is, everyone can (ab)use this by connecting directly to the Asterisk IP. This way they can easi...

...stered in our realm forward(10.10.10.3, 5060); ## Our Cisco router break; }; # retrieve voicemail # if (uri=~"^sip:2[0-9]*@magnum.test.net") { log(1, "Retrieving voicemail\n"); # redirect now! rewritehostport("202.125.25.102:5061"); append_branch(); t_relay_to_udp("202.125.25.106","5061"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404",...

hello all, i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to 1.23 requests from SER to asterisk die quietly, no matter how verbose my asterisk is. It's as if the requests dont exist at all. My setup is as follows: asterisk and SER on the same box, SER running on...

...am trying to redirect the call to the asterisk. --------------------------------------------------------------------------------------------------------- if (method == "INVITE") { if (uri =~ "sip:1[0-9]{4}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xx.xx.xx.xx:xxxx"); t_relay(); break; } } ----------------------------------------------------------------------------------------------------------- inside sip.conf i have -------------------------------------------------------------------------------------...

...f (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "$avp(inv)"); prefix("9"); # route to Asterisk Media Server rewritehostport("192.168.2.75:5060"); append_branch(); t_relay("192.168.2.75:5060"); resetflag(2); } ......................... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara@hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara@...

...# Lookup failed, assume that this is a phone number URL until I setup pattern matching # Send to the Quintum xlog("L_INFO", "Sending URI (%ru) to Quintum"); rewritehostport("192.168.0.147:5060"); }; } # NOTE: I could have combine the following "else if" and "else" but I wanted to seperate them for debug purpo # If it came from a Quintum send it to an Asterisk...

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 requ...

Hi guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf ------------------------------------------------------------------------------------- port=5061 bindaddr=0.0.0.0 srvlookup=yes [ser] type=peer host=xxx.xxx.xxx.xxx...

...some stage but im not sure of the exact sequence or if asterisk contacts ua1 directly or through ser. Somekind of call flow diagrams for this implementation wold be great. Im also trying to implement this in practice. I have ser as a registrar and asterisk set up aswell. I have modifed ser.cfg to rewritehostport(asterisk ip:5061) when not found, however could someone tell me what to modify in my sip.conf,exntensions,voicemail.conf? A simple example if possible please because all the examples I havee seen so far have pstn forwrading implemented also which complicates things. A look at someones working versi...

...to the other party. It seems like the asterisk is not accepting the BYE packet as part of the sip session. I have attached the SIP packets from an ethereal run on the external client side. The same happens if I set the forward to nmibwksip3. If I set it to pbx, the call is not set up. I have tried rewritehostport() instead of forward but this breaks the call setup too. I think that the session state breaks because the asterisk doesn't see the forwarded bye packet as part of the same session. Can I set the name(s) that asterisk answers to, same as the alias statements in ser.cfg? Will that allow me to...

Hello: I noticed in few references that asterisk and ser and complementary. Meaning asterisk handles connections to PSTN and voicemail but SER is better for routing SIP traffic. Is anyone using just asterisk for production purpose. Meaning serving a high number of callers. Is it mandatory to use SER behind asterisk? your feedback would appreciated. -Gaid -------------- next part

...rking Fine. SIP port and RTP ports are forwarded into router to OpenSER System only. openser.cfg listen=192.168.2.11 alias=sip.hyperion.com # Invite Section if ( method== invite ) { # proxy authentication if( uri=="sip:\12345@." ) { # rewriting to Asterisk server for Voicemail messages rewritehostport("192.168.2.75:5060"); exit; } } The Asterisk is executing , But No Audio to the UAC in Behind the NAT .. In Asterisk sever is I'm setting this " chan_Sip.so:[2344] retrans_pkt: Maximum exceeded on transmission " -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkar...

...going out to pstn to my asterisk server. call flow is basically: ua --> opensips server --> * server --> sip gateway provider if (uri=~"sip:00[0-9]*@sip\.myserver\.com") { xlog("L_INFO", "Call to PSTN\n"); #strip(2); #prefix("011"); rewritehostport("20.21.22.23:6050"); <--- IP and Port of * Server route(1); exit; } call routing works properly, but i would like for the rtp not to go thru asterisk, i'm using the canreinvite option, but when i try to make a call, rtp debug still sees rtp passing thru the asterisk. Se...

...sy\n"); } else { if (!subst_user('/^/*voicemail-noanswer-/')){ log(1,"Err in subst_user\n"); } xlog("L_ERR", "Relaying to voicemail No answer\n"); } rewritehostport("FEATURE:5060"); append_branch(); t_relay(); } } >From the gateway's point of view, the invite looks like 1. U GATEWAY:5060 -> PROXY:5060 2. INVITE sip:DSTNUM@PROXY SIP/2.0. 3. Via: SIP/2.0/UDP GATEWAY:5060;branch=z9hG4bK6e117757. 4. From: "SRC...

...t;); break; }; #end # nothing found, try PSTN if (method=="CANCEL" || method=="BYE" || method=="ACK") { xlog("L_INFO", "%is [%Tf]: %rm %fu -> %ru [R4]: just forwarding to PSTN"); rewritehostport("1.2.3.4:5060"); t_relay(); break; }; ====== end of configurations ========

Thank you Melissa. I love the phone but the dial keypad is a little bouncy. I was hoping for a more solid feel like on the analog PT390's or my quality standard, the Nortel 9417CW. Other than the MWI problem, I'd like more documentation on the configuration paramters. I have found little online configuration documentation other than very basic stuff on the Sayson website. I'd