Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I've tried these settings and I still find that I cannot hear the called party. I've also tried what feels like every allow/disallow combination with and without a prefix and I either get 488 errors, using one format when the capability is another errors, or completed calls where I can't hear the called party. So pretty much I feel like I'm just going in circles. Any suggestions? Brad On 20 Mar 2003, Gregg Lebovitz wrote: I remember at some point getting 488 media errors if I didn't enable gsm. Here are my sip.conf and extensions.conf entries. They work for calls out to iconnect: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context=iconnect ; Default for incoming calls disallow=g723.1 [iconnecthere] type=friend username= secret= host=sipauth.deltathree.com context=default disallow=g723.1 allow=gsm allow=ulaw allow=alaw allow=slinear ;;; extensions.conf exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Directory,default exten = t,1,Goto(#,1) ; If they take too long, give up exten = i,1,Playback(invalid) ; That's not valid, try again exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED] exten = _1XX,2,Congestion On Thu, 2003-03-20 at 18:25, Luke Howard wrote: I've found the same. If I make an outgoing call (snom 200 handset), I get about 5 seconds of audio and then it drops out (very occasionally it does work). Incoming calls appear to work, though. -- Executing Goto(SIP/515-Office-143b, iconnecthere-ulaw|91800XXX|1) in new stack -- Goto (iconnecthere-ulaw,91800XXX,1) -- Executing StripMSD(SIP/515-Office-143b, 1) in new stack -- Executing Dial(SIP/515-Office-143b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-960b answered SIP/515-Office-143b -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b -- Got SIP response 480 Temporarily not available back from 213.137.73.178 == Spawn extension (iconnecthere-ulaw, 1800XXX, 2) exited non-zero on 'SIP/515-Office-143b' SIP config is: [general] port=5060 bindaddr=0.0.0.0 context=sip-remote disallow=all allow=ulaw allow=alaw tos=lowdelay tos=184 register = 1XX:[EMAIL PROTECTED] [iconnecthere] type=friend username= password= host=sipauth.deltathree.com context=iconnecthere-ulaw callerid=PADL Software Pty Ltd (XXX) XXX ;txgain = 5.0; ;rxgain = 5.0; inbanddtmf=1 -- Luke P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As I understand it, buying a LineJACK won't suffice if the card's DSP is not actually used. -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
brad, Just to make sure you understand the settings, not using the prefix tells iconnect to use uncompressed codecs. Using sets iconnect into compressed codec mode. I am experience that same problem as you when I try to use the uncompressed mode. I connect, but cannot hear the other party. Using the prefix with the gsm codec works. I am using an internet line jack as FXS. My linejack card is configured to use format=ulaw. Also, are you using a NAT/PAT gateway, or are you connected directly to the internet? Gregg On Sun, 2003-03-30 at 05:22, Brad Bergman wrote: I've tried these settings and I still find that I cannot hear the called party. I've also tried what feels like every allow/disallow combination with and without a prefix and I either get 488 errors, using one format when the capability is another errors, or completed calls where I can't hear the called party. So pretty much I feel like I'm just going in circles. Any suggestions? Brad On 20 Mar 2003, Gregg Lebovitz wrote: I remember at some point getting 488 media errors if I didn't enable gsm. Here are my sip.conf and extensions.conf entries. They work for calls out to iconnect: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context=iconnect ; Default for incoming calls disallow=g723.1 [iconnecthere] type=friend username= secret= host=sipauth.deltathree.com context=default disallow=g723.1 allow=gsm allow=ulaw allow=alaw allow=slinear ;;; extensions.conf exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Directory,default exten = t,1,Goto(#,1) ; If they take too long, give up exten = i,1,Playback(invalid) ; That's not valid, try again exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED] exten = _1XX,2,Congestion On Thu, 2003-03-20 at 18:25, Luke Howard wrote: I've found the same. If I make an outgoing call (snom 200 handset), I get about 5 seconds of audio and then it drops out (very occasionally it does work). Incoming calls appear to work, though. -- Executing Goto(SIP/515-Office-143b, iconnecthere-ulaw|91800XXX|1) in new stack -- Goto (iconnecthere-ulaw,91800XXX,1) -- Executing StripMSD(SIP/515-Office-143b, 1) in new stack -- Executing Dial(SIP/515-Office-143b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-960b answered SIP/515-Office-143b -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b -- Got SIP response 480 Temporarily not available back from 213.137.73.178 == Spawn extension (iconnecthere-ulaw, 1800XXX, 2) exited non-zero on 'SIP/515-Office-143b' SIP config is: [general] port=5060 bindaddr=0.0.0.0 context=sip-remote disallow=all allow=ulaw allow=alaw tos=lowdelay tos=184 register = 1XX:[EMAIL PROTECTED] [iconnecthere] type=friend username= password= host=sipauth.deltathree.com context=iconnecthere-ulaw callerid=PADL Software Pty Ltd (XXX) XXX ;txgain = 5.0; ;rxgain = 5.0; inbanddtmf=1 -- Luke P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As I understand it, buying a LineJACK won't suffice if the card's DSP is not actually used. -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I'm not behind a NAT, but of course behind a firewall (duh). I was even thinking to myself this is very much like what happens with IAX when there is a firewall issue. So having taken care of that, it works great with the same sip.conf settings you have below, and both directions can hear each other with the uncompressed codecs used. The only problem uncompressed is that I get: NOTICE: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 received But everything sounds ok. I haven't tried a very lengthy conversation though. Brad On 30 Mar 2003, Gregg Lebovitz wrote: brad, Just to make sure you understand the settings, not using the prefix tells iconnect to use uncompressed codecs. Using sets iconnect into compressed codec mode. I am experience that same problem as you when I try to use the uncompressed mode. I connect, but cannot hear the other party. Using the prefix with the gsm codec works. I am using an internet line jack as FXS. My linejack card is configured to use format=ulaw. Also, are you using a NAT/PAT gateway, or are you connected directly to the internet? Gregg On Sun, 2003-03-30 at 05:22, Brad Bergman wrote: I've tried these settings and I still find that I cannot hear the called party. I've also tried what feels like every allow/disallow combination with and without a prefix and I either get 488 errors, using one format when the capability is another errors, or completed calls where I can't hear the called party. So pretty much I feel like I'm just going in circles. Any suggestions? Brad On 20 Mar 2003, Gregg Lebovitz wrote: I remember at some point getting 488 media errors if I didn't enable gsm. Here are my sip.conf and extensions.conf entries. They work for calls out to iconnect: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context=iconnect ; Default for incoming calls disallow=g723.1 [iconnecthere] type=friend username= secret= host=sipauth.deltathree.com context=default disallow=g723.1 allow=gsm allow=ulaw allow=alaw allow=slinear ;;; extensions.conf exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Directory,default exten = t,1,Goto(#,1) ; If they take too long, give up exten = i,1,Playback(invalid) ; That's not valid, try again exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED] exten = _1XX,2,Congestion On Thu, 2003-03-20 at 18:25, Luke Howard wrote: I've found the same. If I make an outgoing call (snom 200 handset), I get about 5 seconds of audio and then it drops out (very occasionally it does work). Incoming calls appear to work, though. -- Executing Goto(SIP/515-Office-143b, iconnecthere-ulaw|91800XXX|1) in new stack -- Goto (iconnecthere-ulaw,91800XXX,1) -- Executing StripMSD(SIP/515-Office-143b, 1) in new stack -- Executing Dial(SIP/515-Office-143b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-960b answered SIP/515-Office-143b -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b -- Got SIP response 480 Temporarily not available back from 213.137.73.178 == Spawn extension (iconnecthere-ulaw, 1800XXX, 2) exited non-zero on 'SIP/515-Office-143b' SIP config is: [general] port=5060 bindaddr=0.0.0.0 context=sip-remote disallow=all allow=ulaw allow=alaw tos=lowdelay tos=184 register = 1XX:[EMAIL PROTECTED] [iconnecthere] type=friend username= password= host=sipauth.deltathree.com context=iconnecthere-ulaw callerid=PADL Software Pty Ltd (XXX) XXX ;txgain = 5.0; ;rxgain = 5.0; inbanddtmf=1 -- Luke P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As I understand it, buying a LineJACK won't suffice if the card's DSP is not actually used. -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Luke, here's some information I got back from iconnect: 1) the prefix is not a toggle. It tells iconnects SIP gateway to use compressed codecs. The choices are gsm, g723.1, g729. If you don't use , the gateway will tried to use PCMu/8000 (ulaw?) or PCMa/8000 (alaw?). I can get the gateway to work with g723.1 and gsm, but I can't get it to work with ulaw or alaw. My phone device is a quicknet linecard. The g723.1 format on the linecard does not work with iconnect. If I use it then the audio to and from iconnect is distorted (as if it is using the wrong format or has sampling errors). Gregg On Thu, 2003-03-20 at 18:25, Luke Howard wrote: I've found the same. If I make an outgoing call (snom 200 handset), I get about 5 seconds of audio and then it drops out (very occasionally it does work). Incoming calls appear to work, though. -- Executing Goto(SIP/515-Office-143b, iconnecthere-ulaw|91800XXX|1) in new stack -- Goto (iconnecthere-ulaw,91800XXX,1) -- Executing StripMSD(SIP/515-Office-143b, 1) in new stack -- Executing Dial(SIP/515-Office-143b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-960b answered SIP/515-Office-143b -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b -- Got SIP response 480 Temporarily not available back from 213.137.73.178 == Spawn extension (iconnecthere-ulaw, 1800XXX, 2) exited non-zero on 'SIP/515-Office-143b' SIP config is: [general] port=5060 bindaddr=0.0.0.0 context=sip-remote disallow=all allow=ulaw allow=alaw tos=lowdelay tos=184 register = 1XX:[EMAIL PROTECTED] [iconnecthere] type=friend username= password= host=sipauth.deltathree.com context=iconnecthere-ulaw callerid=PADL Software Pty Ltd (XXX) XXX ;txgain = 5.0; ;rxgain = 5.0; inbanddtmf=1 -- Luke P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As I understand it, buying a LineJACK won't suffice if the card's DSP is not actually used. -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Gregg, 1) the prefix is not a toggle. It tells iconnects SIP gateway to use compressed codecs. The choices are gsm, g723.1, g729. I figured as much. I'm sticking with G.711 as GSM sounds horrible (at least with the snom phones) and the other codecs you mention are patent encumbered. I can get the gateway to work with g723.1 and gsm, but I can't get it to work with ulaw or alaw. My phone device is a quicknet linecard. The problem I'm having appears to be purely a signalling one. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Is there a Record-Route header in the response that comes back from iconnect? Mark On Sun, 23 Mar 2003, Luke Howard wrote: Or maybe we should send an ACK to them -- I need to read the SIP RFC... Tried that, doesn't work. I should add that in my config I'm totally behind NAT, both asterisk and an ATA186 that talks to it. So that may be confounding me in terms of what I'm seeing. I do see the same problem: after a few minutes, the call is dropped (this is using Asterisk patched to ignore 480 Temporarily not available errors). From the log below it _seems_ like iConnectHere is waiting for an acknowledgment to the 480, but you noted that you tried this? It seems to be purely a signalling problem as the call is setup fine between the SIP phone and the gateway (which in this case appeared to be somewhere in Austria...) -- Executing Macro(SIP/515-Office-b8b1, iconnecthere|33145207135|60) in new stack -- Executing Dial(SIP/515-Office-b8b1, SIP/[EMAIL PROTECTED]|60|r) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-ec91 is ringing -- SIP/iconnecthere-ec91 answered SIP/515-Office-b8b1 -- Attempting native bridge of SIP/515-Office-b8b1 and SIP/iconnecthere-ec91 -- Got SIP response 408 Request Timeout back from 213.137.73.178 == Spawn extension (macro-iconnecthere, s, 1) exited non-zero on 'SIP/515-Office-b8b1' in macro 'iconnecthere' == Spawn extension (local, 933145207135, 1) exited non-zero on 'SIP/515-Office-b8b1' -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Mark, I believe there is: Here is the exchange using sip debug. Gregg --- bigcat*CLI sip debug SIP Debugging Enabled -- Executing Dial(Phone/phone0, SIP/[EMAIL PROTECTED]) in new stack Interface is eth0 IP Address is 192.168.4.3 We're at 192.168.4.3 port 39998 Answering with preferred capability 4 Answering with preferred capability 8 Answering with preferred capability 1 Answering with preferred capability 2 10 headers, 10 lines XXX Need to handle Retransmitting XXX: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e Contact: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 202 v=0 o=root 13858 13858 IN IP4 192.168.4.3 s=session c=IN IP4 192.168.4.3 t=0 0 m=audio 39998 RTP/AVP 0 8 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 (no NAT) to 213.137.73.178:5060 -- Called [EMAIL PROTECTED] Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 Call-ID: [EMAIL PROTECTED] From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 Call-ID: [EMAIL PROTECTED] From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=4b857f0d-5ab788dc CSeq: 102 INVITE Proxy-Authenticate: DIGEST realm=deltathree.com, nonce=3e7e6ed6, algorithm=MD5 Content-Length: 0 8 headers, 0 lines XXX Need to handle Retransmitting XXX: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=4b857f0d-5ab788dc Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 213.137.73.178:5060 We're at 192.168.4.3 port 39998 Answering with preferred capability 4 Answering with preferred capability 8 Answering with preferred capability 1 Answering with preferred capability 2 XXX Need to handle Retransmitting XXX: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e Contact: sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username=85904362, realm=deltathree.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=3e7e6ed6, response=b6bab0a7e409d10496cd6140e6d1e063 Content-Type: application/sdp Content-Length: 202 v=0 o=root 13837 13837 IN IP4 192.168.4.3 s=session c=IN IP4 192.168.4.3 t=0 0 m=audio 39998 RTP/AVP 0 8 4 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 (no NAT) to 213.137.73.178:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 Call-ID: [EMAIL PROTECTED] From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED] CSeq: 103 INVITE Content-Length: 0 7 headers, 0 lines Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0;received=66.30.28.60 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=1FD16250-8D Date: Mon, 24 Mar 2003 02:35:05 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 180 v=0 o=CiscoSystemsSIP-GW-UserAgent 3626 3613 IN IP4 213.137.65.239 s=SIP Call c=IN IP4 213.137.65.239 t=0 0 m=audio 18358 RTP/AVP 4 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no 12 headers, 8 lines Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0;received=66.30.28.60 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=1FD16250-8D Date: Mon, 24 Mar 2003 02:35:05 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:213.137.79.80, sip:213.137.79.78, sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176 Content-Type: application/sdp Content-Length: 180 v=0 o=CiscoSystemsSIP-GW-UserAgent 3626 3613 IN IP4 213.137.65.239 s=SIP Call c=IN IP4 213.137.65.239 t=0 0 m=audio 18358 RTP/AVP 4 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no 14 headers, 8 lines XXX Need to handle Retransmitting XXX: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.3:5060;branch=0ed5cff0 From: asterisk sip:[EMAIL PROTECTED];tag=67ff402e To: sip:[EMAIL PROTECTED];tag=4b857f0d-5ab788dc Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 213.137.73.178:5060 --
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I remember at some point getting 488 media errors if I didn't enable gsm. As I mentioned, I'm getting 480 Temporarily not available, not 488 media errors. I tried the grotesque hack of making handle_response() ignore 480 errors, which *seems* to work. Hmm. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Luke Howard wrote:I remember at some point getting 488 media errors if I didn't enable gsm. As I mentioned, I'm getting 480 Temporarily not available, not 488 media errors. I tried the grotesque hack of making handle_response() ignore 480 errors, which *seems* to work. Hmm. I tried that, and at least for me it has a number of subtle side effects: 1. Calls all cut off after just a few minutes 2. Subsequent calls after those messages have been ignored are bollixed up. I have complained repeatedly to iconnecthere, but they don't have much of a customer service model. Canned email responses is about the long and short of it. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I tried the grotesque hack of making handle_response() ignore 480 errors, which *seems* to work. Hmm. I tried that, and at least for me it has a number of subtle side effects: Or maybe we should send an ACK to them -- I need to read the SIP RFC... -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Moreover, if anyone has a packet trace of iConnectHere's SIP client making a call (which presumably does work), then please send it along... it would be interesting to see whether Asterisk, at fault or not, can be made to work around this properly. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Luke Howard wrote:I tried the grotesque hack of making handle_response() ignore 480 errors, which *seems* to work. Hmm. I tried that, and at least for me it has a number of subtle side effects: Or maybe we should send an ACK to them -- I need to read the SIP RFC... Tried that, doesn't work. I should add that in my config I'm totally behind NAT, both asterisk and an ATA186 that talks to it. So that may be confounding me in terms of what I'm seeing. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I should add that in my config I'm totally behind NAT, both asterisk and an ATA186 that talks to it. Hmm, both our SIP phones and Asterisk are on visible IPs. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Or maybe we should send an ACK to them -- I need to read the SIP RFC... Tried that, doesn't work. I should add that in my config I'm totally behind NAT, both asterisk and an ATA186 that talks to it. So that may be confounding me in terms of what I'm seeing. I do see the same problem: after a few minutes, the call is dropped (this is using Asterisk patched to ignore 480 Temporarily not available errors). From the log below it _seems_ like iConnectHere is waiting for an acknowledgment to the 480, but you noted that you tried this? It seems to be purely a signalling problem as the call is setup fine between the SIP phone and the gateway (which in this case appeared to be somewhere in Austria...) -- Executing Macro(SIP/515-Office-b8b1, iconnecthere|33145207135|60) in new stack -- Executing Dial(SIP/515-Office-b8b1, SIP/[EMAIL PROTECTED]|60|r) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-ec91 is ringing -- SIP/iconnecthere-ec91 answered SIP/515-Office-b8b1 -- Attempting native bridge of SIP/515-Office-b8b1 and SIP/iconnecthere-ec91 -- Got SIP response 408 Request Timeout back from 213.137.73.178 == Spawn extension (macro-iconnecthere, s, 1) exited non-zero on 'SIP/515-Office-b8b1' in macro 'iconnecthere' == Spawn extension (local, 933145207135, 1) exited non-zero on 'SIP/515-Office-b8b1' -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
GSM works but the voice quality is absolutely terrible. This is the case with or without the prefix. (Did anyone ever figure out whether is a toggle?) One thing I didn't realise until reading the new documentation is that the codec list is in order of preference. So, if there's an advantage to advertising GSM without actually employing it (can't imagine why, but...) then this is how to do it. Interestingly, iConnectHere seems to be letting me make calls now. It does seem rather tepermental! At least incoming calls appear to work, I can always make outgoing calls over the local PSTN... -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I've found the same. If I make an outgoing call (snom 200 handset), I get about 5 seconds of audio and then it drops out (very occasionally it does work). Incoming calls appear to work, though. -- Executing Goto(SIP/515-Office-143b, iconnecthere-ulaw|91800XXX|1) in new stack -- Goto (iconnecthere-ulaw,91800XXX,1) -- Executing StripMSD(SIP/515-Office-143b, 1) in new stack -- Executing Dial(SIP/515-Office-143b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-960b answered SIP/515-Office-143b -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b -- Got SIP response 480 Temporarily not available back from 213.137.73.178 == Spawn extension (iconnecthere-ulaw, 1800XXX, 2) exited non-zero on 'SIP/515-Office-143b' SIP config is: [general] port=5060 bindaddr=0.0.0.0 context=sip-remote disallow=all allow=ulaw allow=alaw tos=lowdelay tos=184 register = 1XX:[EMAIL PROTECTED] [iconnecthere] type=friend username= password= host=sipauth.deltathree.com context=iconnecthere-ulaw callerid=PADL Software Pty Ltd (XXX) XXX ;txgain = 5.0; ;rxgain = 5.0; inbanddtmf=1 -- Luke P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As I understand it, buying a LineJACK won't suffice if the card's DSP is not actually used. -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Luke, Try putting the prefix of before your phone number. It changes the codec expected by iconnect. Gregg On Thu, 2003-03-20 at 18:25, Luke Howard wrote: I've found the same. If I make an outgoing call (snom 200 handset), I get about 5 seconds of audio and then it drops out (very occasionally it does work). Incoming calls appear to work, though. -- Executing Goto(SIP/515-Office-143b, iconnecthere-ulaw|91800XXX|1) in new stack -- Goto (iconnecthere-ulaw,91800XXX,1) -- Executing StripMSD(SIP/515-Office-143b, 1) in new stack -- Executing Dial(SIP/515-Office-143b, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-960b answered SIP/515-Office-143b -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b -- Got SIP response 480 Temporarily not available back from 213.137.73.178 == Spawn extension (iconnecthere-ulaw, 1800XXX, 2) exited non-zero on 'SIP/515-Office-143b' SIP config is: [general] port=5060 bindaddr=0.0.0.0 context=sip-remote disallow=all allow=ulaw allow=alaw tos=lowdelay tos=184 register = 1XX:[EMAIL PROTECTED] [iconnecthere] type=friend username= password= host=sipauth.deltathree.com context=iconnecthere-ulaw callerid=PADL Software Pty Ltd (XXX) XXX ;txgain = 5.0; ;rxgain = 5.0; inbanddtmf=1 -- Luke P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As I understand it, buying a LineJACK won't suffice if the card's DSP is not actually used. -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iconnecthere 480 error: is there a workaround?
I am going to have to find a fix for this problem or I'm going to have to quit using iconnect. About one call in 10 or so, iconnect's gateway gives me an error (console output appended below). So upon receiving the error, which as a 4XX error means, Fatal, asterisk gives up and drops the call. But not iconnect!! The phone at the other end starts ringing, and rings several times before the call is dropped. So the person at the other end, unless it's my friends who are now inured to this, wonder WTF is going on. I sent a mail to iconnect asking if they don't agree that it's broken, but in the near-term I need to find a fix. Thx. B. * Console output begins here, numbers elided to protect the innocent :-) -- Called [EMAIL PROTECTED] -- Got SIP response 480 Temporarily not available back from 213.137.73.140 == No one is available to answer at this time WARNING[311310]: File pbx.c, Line 1179 (ast_pbx_run): Channel 'SIP/ata1-2da9' sent into invalid extension '1XXXNNN' in context 'iconn', but no invalid handler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
