<html><head></head><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:16px"><div id="yui_3_16_0_1_1453291685653_13573" class="">Hello;</div><div id="yui_3_16_0_1_1453291685653_13573" class=""><br></div><div id="yui_3_16_0_1_1453291685653_13573" class="">Thanks a lot for your kindly reply.</div><div id="yui_3_16_0_1_1453291685653_13573" class="">Actually the alaw is enabled at asterisk but what I got to know from the other side that they only enabled ulaw. Below is my asterisk sip configuration for the sip trunk. Please advise.</div><div id="yui_3_16_0_1_1453291685653_13573" class=""><br></div><div id="yui_3_16_0_1_1453291685653_13573" class="">[user_name]</div><div id="yui_3_16_0_1_1453291685653_13573" class="">type=peer</div><div id="yui_3_16_0_1_1453291685653_13573" class="">host=Provider_IP_Address</div><div id="yui_3_16_0_1_1453291685653_13573" class="">port=5083</div><div id="yui_3_16_0_1_1453291685653_13573" class="">context=trunkinbound</div><div id="yui_3_16_0_1_1453291685653_13573" class="">disallow=all</div><div id="yui_3_16_0_1_1453291685653_13573" class="">allow = ulaw,alaw,gsm</div><div id="yui_3_16_0_1_1453291685653_13573" class="">call-limit = 256 </div><div id="yui_3_16_0_1_1453291685653_13573" class="">insecure = port,invite</div><div id="yui_3_16_0_1_1453291685653_13573" class="">trunkstyle = provider</div><div id="yui_3_16_0_1_1453291685653_13573" class="">transport = udp </div><div id="yui_3_16_0_1_1453291685653_13573" class="">dtmfmode = rfc2833</div><div id="yui_3_16_0_1_1453291685653_13573" class="">remoteregister = yes</div><div id="yui_3_16_0_1_1453291685653_13573" class="">cbcallerid = 22021782</div><div id="yui_3_16_0_1_1453291685653_13573" class="">qualify = yes</div><div id="yui_3_16_0_1_1453291685653_13573"></div><div id="yui_3_16_0_1_1453291685653_13573" dir="ltr" class="">srtpcapable = no</div><div id="yui_3_16_0_1_1453291685653_13573" dir="ltr" class=""><br></div><div id="yui_3_16_0_1_1453291685653_13573" dir="ltr" class="">Regards</div><div id="yui_3_16_0_1_1453291685653_13573" dir="ltr" class="">Bilal</div> <div class="qtdSeparateBR"><br><br></div><div class="yahoo_quoted" style="display: block;"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;"> <div dir="ltr"><font size="2" face="Arial"> On Wednesday, January 20, 2016 2:50 PM, A J Stiles <asterisk_list@earthshod.co.uk> wrote:<br></font></div> <br><br> <div class="y_msg_container">On Wednesday 20 Jan 2016, bilal ghayyad wrote:<br clear="none">> Hello List;<br clear="none">> I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and<br clear="none">> I am getting the following debug, can someone advise me about the<br clear="none">> solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE<br clear="none">> ..... [stuff deleted] .....<div class="yqt5639131691" id="yqtfd66723"><br clear="none">> [Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37]<---<br clear="none">> Reliably Transmitting (no NAT) to Provider_IP_Address:5083 --->SIP/2.0 488<br clear="none">> Not acceptable here Via: SIP/2.0/UDP<br clear="none">> Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1;received=Pro<br clear="none">> vider_IP_Address From: "1828444" <sip:<a shape="rect" ymailto="mailto:1828444@c4.gw" href="mailto:1828444@c4.gw">1828444@c4.gw</a>>;tag=rrZpHF51Z7a6D To:<br clear="none">> <sip:<a shape="rect" ymailto="mailto:22021782@Asterisk_IP_Address" href="mailto:22021782@Asterisk_IP_Address">22021782@Asterisk_IP_Address</a>:5060>;tag=as5d16dbaf Call-ID:<br clear="none">> 6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFq<br clear="none">> loi5 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL,<br clear="none">> OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported:<br clear="none">> replaces Content-Length: 0 <------------></div><br clear="none"><br clear="none">"488 Not acceptable here" usually means that negotiation failed for want of <br clear="none">any mutually-supported codec. Make sure that you have "alaw", which is the <br clear="none">native format used by the PSTN in civilised countries (and therefore, there <br clear="none">is little need to use anything else unless you know you will never want PSTN <br clear="none">connectivity), enabled at your end.<br clear="none"><br clear="none"><br clear="none">Can you run this command and post the output? (It should all be on one line, <br clear="none">but my mail client or yours may have eaten it)<br clear="none"><br clear="none">$ awk '/[[]|allow/&&!/^[ \t]*;/{printf "%6d:%s\n",NR, $0}' <br clear="none">/etc/asterisk/sip.conf<br clear="none"><br clear="none">This will look for [section headers] in square brackets and lines containing <br clear="none">"allow" (which also will catch "disallow") that are not commented out, in your <br clear="none">SIP configuration, and print them out with line numbers.<br clear="none"><br clear="none"><br clear="none">-- <br clear="none">AJS<br clear="none"><br clear="none">Note: Originating address only accepts e-mail from list! If replying off-<br clear="none">list, change address to asterisk1list at earthshod dot co dot uk .<div class="yqt5639131691" id="yqtfd69368"><br clear="none"></div><br><br></div> </div> </div> </div></div></body></html>