[Asterisk-Users] Re: Polycom randomly fails outbound calls,
Noah Miller wrote: Hi Andres - The two that we have are just used as lobby phones. They're good little phones, but if you have the money, I'd definitely recommend the IP501 instead. The screen is MUCH better, and having full speakerphone is great! Plus the 500/501 just feels a little more solid. Yeah, I think it was a wrong move going for the 301 instead, Hmm. I'm not sure either. I've never used AMP before (except for a quick glance at [EMAIL PROTECTED]). If you can change the sip settings, I don't think it should matter. I am not using [EMAIL PROTECTED], it didn't work well for me, I just using AMP over asterisk, and yes, sip are 100% tweakable, how do you configure your system, all by hand? Well, the two weird things I see here are the type setting and the host. Type is set to peer, but there doesn't seem to be a corresponding user definition (AFAIK, all peers have to have users). You might try changing it to type=friend instead (like 201). I did it, it was set to peer just because I red somewhere that Polys didn't like friend type, For the host setting, this is the address of the sip device, and not the asterisk server. If you have the Polycom set to a static address of 192.168.1.18, all is well. If your Polycom is set to DHCP (this is the default), you should use host=dynamic it's fixed to 18 A couple of things that I know you don't need: nat=never qualify=no I took them off too, I got them from the only how to I found about amp and the polycom, - Noah Thanks , I hope I can help you same day, Andres ___ --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] Re: Polycom randomly fails outbound calls,
Hi Andres - I am not using [EMAIL PROTECTED], it didn't work well for me, I just using AMP over asterisk, and yes, sip are 100% tweakable, how do you configure your system, all by hand? Yeah, by hand. When I first started doing this there was no such thing as AMP. Plus, I've got some wacky dialplan stuff that probably wouldn't work out too well with AMP. Well, the two weird things I see here are the type setting and the host. Type is set to peer, but there doesn't seem to be a corresponding user definition (AFAIK, all peers have to have users). You might try changing it to type=friend instead (like 201). I did it, it was set to peer just because I red somewhere that Polys didn't like friend type, For the host setting, this is the address of the sip device, and not the asterisk server. If you have the Polycom set to a static address of 192.168.1.18, all is well. If your Polycom is set to DHCP (this is the default), you should use host=dynamic it's fixed to 18 Well, if you've got all this stuff and the phones still aren't working, I'd say there's either something funny going on with AMP or Asterisk. I would try new/different versions of both. The rest of your configs look good to me. Maybe somebody else can spot an inconsistency? Just to cover all the bases, have you tried configuring the Polycom without AMP? - Noah ___ --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] Re: Polycom randomly fails outbound calls,
Hi Andres - I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301 The Polycom misses 1 out of 2 dialout calls, this is the full log from a call which didn't go through. 303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found 303095 Sep 14 10:45:15 DEBUG[15427]: Ooh, format changed from unknown to ulaw 303096 Sep 14 10:45:15 DEBUG[15427]: Didn't get a frame from channel: SIP/pstn_2-1f35 303097 Sep 14 10:45:15 DEBUG[15427]: Bridge stops bridging channels SIP/200-0db1 and SIP/pstn_2-1f35 I guess the line 303096 is the more relevant, but I don't know where to start troubleshooting it. Line 303095 is probably relevant, too. What codec is the phone configured to try first? It looks like the phone is trying to use something asterisk doesn't understand, or is not configured for. Maybe set the phone to ulaw instead. Also, what dtmfmode are you using? Can we look at your sip.conf from asterisk, and the config files for your Polycom phone? - Noah ___ --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] Re: Polycom randomly fails outbound calls,
More information in this thread, This Poly 301 sometimes rings out sometimes doesn't, it calls out through * using spa3000's fxo, I got this log 1129 Sep 14 15:03:55 DEBUG[15702]: Outgoing Call for ww9821642 1130 Sep 14 15:03:55 DEBUG[15702]: ww9821642 is not a local user 1131 Sep 14 15:03:55 VERBOSE[15702]: -- Called pstn_2/ww9821642 1132 Sep 14 15:03:55 DEBUG[15690]: Acked pending invite 102 1133 Sep 14 15:03:55 DEBUG[15690]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found 1134 Sep 14 15:03:55 DEBUG[15690]: (Provisional) Stopping retransmission (but retaining packet) on '1cd1aa2160ed236e5e5db09e2e7916d[EMAIL PROTECTED]' Request 103: Found 1135 Sep 14 15:03:55 DEBUG[15690]: (Provisional) Stopping retransmission (but retaining packet) on '1cd1aa2160ed236e5e5db09e2e7916d[EMAIL PROTECTED]' Request 103: Found 1136 Sep 14 15:03:55 VERBOSE[15702]: -- SIP/pstn_2-4fc7 is ringing 1137 Sep 14 15:03:56 DEBUG[15690]: Acked pending invite 103 1138 Sep 14 15:03:56 DEBUG[15690]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 103: Found 1139 Sep 14 15:03:56 DEBUG[15690]: build_route: Contact hop: pstn spa 3002 sip:[EMAIL PROTECTED]:5061 1140 Sep 14 15:03:56 VERBOSE[15702]: -- SIP/pstn_2-4fc7 answered SIP/200-eb9f 1141 Sep 14 15:03:56 VERBOSE[15702]: -- Attempting native bridge of SIP/200-eb9f and SIP/pstn_2-4fc7 1142 Sep 14 15:03:56 DEBUG[15702]: Ooh, format changed from unknown to ulaw 1143 Sep 14 15:03:56 DEBUG[15690]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found 1144 Sep 14 15:03:56 DEBUG[15702]: Ooh, format changed from unknown to ulaw 1145 Sep 14 15:04:02 DEBUG[15690]: Auto destroying call '[EMAIL PROTECTED]' 1146 Sep 14 15:04:02 DEBUG[15690]: Auto destroying call '[EMAIL PROTECTED]' 1147 Sep 14 15:04:23 DEBUG[15702]: Didn't get a frame from channel: SIP/200-eb9f 1148 Sep 14 15:04:23 DEBUG[15702]: Bridge stops bridging channels SIP/200-eb9f and SIP/pstn_2-4fc7 1149 Sep 14 15:04:23 DEBUG[15702]: update_user_counter(ww9821642) - decrement outUse counter 1150 Sep 14 15:04:23 DEBUG[15702]: ww9821642 is not a local user 1151 Sep 14 15:04:23 DEBUG[15702]: Exiting with DIALSTATUS=ANSWER. 1152 Sep 14 15:04:23 VERBOSE[15702]: == Spawn extension (macro-dialout-trunk, s, 17) exited non-zero on 'SIP/200-eb9f' in macro 'dialout-trunk' 1153 Sep 14 15:04:23 VERBOSE[15702]: == Spawn extension (from-internal, 9821642, 1) exited non-zero on 'SIP/200-eb9f' 1154 Sep 14 15:04:23 VERBOSE[15702]: -- Executing Macro(SIP/200-eb9f, hangupcall) in new stack 1155 Sep 14 15:04:23 VERBOSE[15702]: -- Executing ResetCDR(SIP/200-eb9f, w) in new stack 1156 Sep 14 15:04:23 VERBOSE[15702]: -- Executing NoCDR(SIP/200-eb9f, ) in new stack I am forcing all, the sipura and the polycom to use only ulaw, but still I get that 'unknown' protocol, and no frames, nor from the polycom, nor from the spa Also this happends randomly there's nothing that gives a pattern, I am using rfc2833 as dtmf mode I already tweaked the dialplan.digitmap= (to an empty string) so everything gets out. my phone's sip.cfg codec setting looks like preferences voice.codecPref.G711Mu=1 voice.codecPref.G711A=2 voice.codecPref.G729AB=3 voice.codecPref.IP_4000.G711Mu=1 voice.codecPref.IP_4000.G711A=2 voice.codecPref.IP_4000.G729AB=/ And ulaw is set as preferred in the extension. as shown in sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown Thanks again, ___ --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] Re: Polycom randomly fails outbound calls,
Thanks Noah for your time, I am using rfc2833 as dtmf mode I already tweaked the dialplan.digitmap= (to an empty string) so everything gets out. my phone's sip.cfg codec setting looks like preferences voice.codecPref.G711Mu=1 voice.codecPref.G711A=2 voice.codecPref.G729AB=3 voice.codecPref.IP_4000.G711Mu=1 voice.codecPref.IP_4000.G711A=2 voice.codecPref.IP_4000.G729AB=/ And ulaw is set as preferred in the extension. as shown in sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown Thanks again, Andres -- I guess the line 303096 is the more relevant, but I don't know where to start troubleshooting it. Line 303095 is probably relevant, too. What codec is the phone configured to try first? It looks like the phone is trying to use something asterisk doesn't understand, or is not configured for. Maybe set the phone to ulaw instead. Also, what dtmfmode are you using? Can we look at your sip.conf from asterisk, and the config files for your Polycom phone? ___ --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
