[Asterisk-Users] Unable to Register to Asterisk through Proxy
Hi, Has anybody come across a situation where they were unable to register with Asterisk through a SIP stateless proxy server? I'm getting an error: 403 Authentication user name does not match account name As far as I can tell the requests reaching Asterisk with and without the proxy are identical excepting the IP address the REGISTER request is coming from and the Via header (nonces and digests are different of course but I've verified that the md5 hash is correct). As far as the information you'd expect to be used for the REGISTER operation I can't see any difference. Success (no proxy) and failure (with proxy) logs for the REGISTER request are below. INVITE requests through the same proxy work correctly with the same the same credentials. It just seems to be some IP address matching going on for the REGISTER command that's causing a problem. I have had a look at chan_sip.c but haven't been able to work it out as of yet. Thanks, Aaron sip.conf for user test [test] type=friend host=dynamic nat=yes canreinvite=no username=test secret=test == Failure REGISTER through Proxy: xxx.xxx.xxx.xxx = Asterisk yyy.yyy.yyy.yyy = Proxy zzz.zzz.zzz.zzz = User Agent Public IP 192.168.1.2 = User Agent Private IP -- SIP read from yyy.yyy.yyy.yyy:5060: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKCefqJfo5hAO/paxgvw/iR7owic4= Via: SIP/2.0/UDP 192.168.1.2:5066;received=zzz.zzz.zzz.zzz:64073;rport=64073;branch=z9hG4bK882020b45b From: test sip:[EMAIL PROTECTED];tag=63274377059065 To: test sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 65 REGISTER Max-Forwards: 69 Expires: 600 Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 0: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 (36) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 1: Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKCefqJfo5hAO/paxgvw/iR7owic4= (80) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 2: Via: SIP/2.0/UDP 192.168.1.2:5066;received=zzz.zzz.zzz.zzz:64073;rport=64073;branch=z9hG4bK882020b45b (101) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 3: From: test sip:[EMAIL PROTECTED];tag=63274377059065 (62) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 4: To: test sip:[EMAIL PROTECTED] (37) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 5: Contact: sip:[EMAIL PROTECTED] (35) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 6: Call-ID: [EMAIL PROTECTED] (36) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 7: CSeq: 65 REGISTER (17) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 8: Max-Forwards: 69 (16) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 9: Expires: 600 (12) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3106 sip_alloc: Allocating new SIP dialog for [EMAIL PROTECTED] - REGISTER (No RTP) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:10945 handle_request: Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to yyy.yyy.yyy.yyy : 5060 (non-NAT) Transmitting (NAT) to yyy.yyy.yyy.yyy:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKCefqJfo5hAO/paxgvw/iR7owic4=;received=yyy.yyy.yyy.yyy Via: SIP/2.0/UDP 192.168.1.2:5066;received=zzz.zzz.zzz.zzz:64073;rport=64073;branch=z9hG4bK882020b45b From: test sip:[EMAIL PROTECTED];tag=63274377059065 To: test sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 65 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to yyy.yyy.yyy.yyy:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKCefqJfo5hAO/paxgvw/iR7owic4=;received=yyy.yyy.yyy.yyy Via: SIP/2.0/UDP 192.168.1.2:5066;received=zzz.zzz.zzz.zzz:64073;rport=64073;branch=z9hG4bK882020b45b From: test sip:[EMAIL PROTECTED];tag=63274377059065 To: test sip:[EMAIL PROTECTED];tag=as738d9ccd Call-ID: [EMAIL PROTECTED] CSeq: 65 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=6a62f137 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms buffalo*CLI -- SIP read from yyy.yyy.yyy.yyy:5060: REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKaN3LqUmNnA6Bm9VM4+7lrnh97Bo= Via: SIP/2.0/UDP 192.168.1.2:5066;received=zzz.zzz.zzz.zzz:64073;rport=64073;branch=z9hG4bKe494c5046b From: test
RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold
-Original Message- From: Pedro Nunes [mailto:[EMAIL PROTECTED] Sent: 15 December 2005 08:59 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold Hello, Do you try Answer() and then Dial(SIP/xyz,,m)??? Exten = ???,1,Answer() Exten = ???,2,Dial(SIP/xyz,,m) You need to answer the call before you can hear music on hold. Hi Pedro, What you suggest would work but is no good as anybody calling our numbers would be charged for the call. The Dial(,,m) command can play MusicOnHold without answering the call, I know I've tested it ;-). In this case I just need to give the RTP a kick start or something, the console reports the MusicOnHold has started playing but there is no RTP. Thanks, Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold
-Original Message- From: Elton Machado [mailto:[EMAIL PROTECTED] Sent: 15 December 2005 14:03 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Starting RTP with Dial and MusicOnHold Why not to use r option in Dial(SIP/xyz,,r) to simulate the ring? Regards, Hi Elton, Tried that one as well. The Dial(,,r) command actually does the opposite of what I want. The r option specifies that no audio, i.e. no RTP stream, should be passed until the call is answered. This option will generate a SIP 180 Ringing response on an incoming call but since in this case the Cerpack switch needs out of band signalling any 180, 183 or other SIP repsonses are ignored for call progress indication. Thanks, Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Exceptionally long queue in SIP Channel
Hi, Started getting a bombardment of these messages on the Asterisk console this morning (20+ a second): Dec 14 10:00:30 WARNING[17006]: channel.c:588 ast_queue_frame: Exceptionally long queue length queuing to SIP/bluecity29-a5cfDec 14 10:00:30 WARNING[17006]: channel.c:603 ast_queue_frame: Unable to write to alert pipe on SIP/bluecity29-a5cf, frametype/subclass 5/0 (qlen = 173842): Resource temporarily unavailable! Had to restart Asterisk to get rid of them. Has anybody seen this before? Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Starting RTP with Dial and MusicOnHold
Hi, I'm trying to get Asterisk working with a supplier's Cerpack switch and everything is working except audio ringback for calls coming from Cerpack to Asterisk. The Cerpack switch only does out of band progress indication (seems a bit strange for SIP to SIP calls?!) so I've spent the last two days trying to find a way to force Asterisk to send an RTP stream to Cerpack for ring back. Theoretically the Dial command with the m option looks to be exactly what I need: Dial(SIP/xyz,,m) This should play musiconhold back to the caller and in my case I just took a recording of the PSTN tones I wanted to play and created a musiconhold class for them. The command will work correctly when dialled from a SIP phone connected to Asterisk but not for calls coming from Cerpack. As far as I can tell this is because Asterisk won't initiate the RTP stream and waits for a packet from the client before starting to play the musiconhold, perhaps assuming the connection is not available until it gets a packet. In this case Cerpack isn't sending a packet so no audio is heard until the call is answered. Has anybody seen anything like this before? Thanks, Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue?? (Solved)
Hi, I got the person to force the G729 codec on their Linksys WRT54GP2 and forced it on Asterisk as well. The person then managed to get a single call out but all subsequent call set ups failed with the same 488 error. I went back over my SIP traces and noticed that the Cseq's were often out of order or duplicated. This looked a lot more like the cause and was more inline with a timing issue which would explain why it was only happening over satellite. I did some more digging and came across the SIP timing settings defined in the SIP RFC. I didn't get a chance too read exactly the mecahnism but one of these settings does seem to be the interval between resending INVITE requests. The good news for me and anybody else reading this is with the same problem is that changing the SIP T1 parameter does get the INVITE requests through. It's on the SIP configuration page for the Linksys/Sipura devices. In this case it was changed from the default 0.5s to 2s and then finally to 4s after which outgoing call set up reliably worked. In addition it does look like there is a bug in the Asterisk SIP channel possibly to do with getting confused about receiving a bunch of INIVTE requests with the same Cseq and stale nonces. It could be related to the recent 403 problem for the Asterisk SIP channel and the Sipura REGISTER requests with stale nonces. I will attempt to replicate the SIP dialogue and produce a SIP trace and if successful file a bug report. Aaron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Clauson Sent: 24 November 2005 03:07 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection,SIP Timing Issue?? Hi, Thanks for the tip I'll try it out. That would explain some situations where one of the peeople concerned was mucking around with the codec settings on the PAP2 and managed to get some calls out. It's a bit baffling how the Linksys devices will get INVITES through without G.729 being set across non-satellite links and yet can't get the very same INVITE through across a satellite link. Fair enough if it was the Linksys generating the 488 during the INVITE negotiation but how does Asterisk even know the difference?? Aaron -Original Message- From: Jason p [mailto:[EMAIL PROTECTED] Sent: 24 November 2005 02:25 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue?? I had the same problem when we were setting up these boxes after katrina. What i found is that they will only do one G729 session at a time. so that mesg that your showing is that its trying to register two chans as 729. what i did to get around this was to turn off fource prefered codec on one line. This threw me for a loop also but trust me this is the fix, and yes you can only make one 729 call at a time. Jason Price On 11/23/05, Aaron Clauson [EMAIL PROTECTED] wrote: Hi, I have a very strange Asterisk SIP call signalling problem that is proving extremely difficult to track down. The problem is that any SIP INVITE request that is coming into Asterisk over a satellite connection from a Linksys Router or PAP2 is getting a Not Acceptable Here (codec error) from Asterisk. I've done all the normal checks on the allowed codecs in sip.conf but to no avail. I've even gone as far as writing a basic SIP stack to authenticate and send the INVITE request to Asterisk with exactly the same SDP payload to let me brute force different options in the SDP request to try an narrow it down that way. The preplexing thing from that length exercise is that if exactly the same INVITE request comes in from my app across the same satellite connection to Asterisk it gets 200 Ok'ed but coming from the Linksys PAP2 or WRT54GP2 it gets 488 Codec Not Acceptable Here'ed. The first time this happened we went through all the usual checks and got nowhere and the person drifted off and it was put down to something speicifc to that set up/connection. But now it's cropped up again with a different person who also just happens to be on a satellite connection but from a different provider, although it is possible both providers use the same infrastructure. In both cases incoming calls to the Linksys devices worked correctly it's just the outgoing calls from the devices to Asterisk that are getting the rejection. In the second case we can't put it down to something to do with the connection because the person has a Vonage service working no problems across the same satellite link we are getting
[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??
Hi, I have a very strange Asterisk SIP call signalling problem that is proving extremely difficult to track down. The problem is that any SIP INVITE request that is coming into Asterisk over a satellite connection from a Linksys Router or PAP2 is getting a Not Acceptable Here (codec error) from Asterisk. I've done all the normal checks on the allowed codecs in sip.conf but to no avail. I've even gone as far as writing a basic SIP stack to authenticate and send the INVITE request to Asterisk with exactly the same SDP payload to let me brute force different options in the SDP request to try an narrow it down that way. The preplexing thing from that length exercise is that if exactly the same INVITE request comes in from my app across the same satellite connection to Asterisk it gets 200 Ok'ed but coming from the Linksys PAP2 or WRT54GP2 it gets 488 Codec Not Acceptable Here'ed. The first time this happened we went through all the usual checks and got nowhere and the person drifted off and it was put down to something speicifc to that set up/connection. But now it's cropped up again with a different person who also just happens to be on a satellite connection but from a different provider, although it is possible both providers use the same infrastructure. In both cases incoming calls to the Linksys devices worked correctly it's just the outgoing calls from the devices to Asterisk that are getting the rejection. In the second case we can't put it down to something to do with the connection because the person has a Vonage service working no problems across the same satellite link we are getting the rejection on. The SIP trace is below and I'm wondering if anybody has ever seen something similar. The only thing I can think of is that it's somehow a timing issue I can't see how it can be a codec issue since the exactly the same SDP payload will get OK'ed if coming from my app. Is the Asterisk SIP stack sensitive to the any timings in the INVITE request? It seems highly unlikely but I just can't think of anything else. INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f From: XXX sip:[EMAIL PROTECTED];tag=831f2cca367c3ddfo1 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=XXX,realm=asterisk,nonce=489bfe04,uri=sip:[EMAIL PROTECTED],al gorithm=MD5,response=22f566e03a225047469d73bec5ab640c Contact: XXX sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 424 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 418210 418210 IN IP4 192.168.1.248 s=- c=IN IP4 192.168.1.248 t=0 0 m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f;received=xxx;rport=5061 From: xxx sip:[EMAIL PROTECTED];tag=831f2cca367c3ddfo1 To: sip:[EMAIL PROTECTED];tag=as17d663fb Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=48554be3 Content-Length: 0 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-c341696b From: xxx sip:[EMAIL PROTECTED];tag=831f2cca367c3ddfo1 To: sip:[EMAIL PROTECTED];tag=as50c8f92d Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username=xxx,realm=asterisk,nonce=3cb4e5eb,uri=sip:[EMAIL PROTECTED],al gorithm=MD5,response=d4438aec627cefa82b6388a3b0c2cb1f Contact: xxx sip:[EMAIL PROTECTED]:5061 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 0 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f From: xxx sip:[EMAIL PROTECTED];tag=831f2cca367c3ddfo1 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=xxx,realm=asterisk,nonce=489bfe04,uri=sip:[EMAIL PROTECTED],al gorithm=MD5,response=22f566e03a225047469d73bec5ab640c Contact: xxx sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 424 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 418210 418210 IN IP4 192.168.1.248 s=- c=IN IP4 192.168.1.248 t=0 0 m=audio 16450 RTP/AVP 0 2 4 8 18
RE: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??
Hi, Thanks for the tip I'll try it out. That would explain some situations where one of the peeople concerned was mucking around with the codec settings on the PAP2 and managed to get some calls out. It's a bit baffling how the Linksys devices will get INVITES through without G.729 being set across non-satellite links and yet can't get the very same INVITE through across a satellite link. Fair enough if it was the Linksys generating the 488 during the INVITE negotiation but how does Asterisk even know the difference?? Aaron -Original Message- From: Jason p [mailto:[EMAIL PROTECTED] Sent: 24 November 2005 02:25 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue?? I had the same problem when we were setting up these boxes after katrina. What i found is that they will only do one G729 session at a time. so that mesg that your showing is that its trying to register two chans as 729. what i did to get around this was to turn off fource prefered codec on one line. This threw me for a loop also but trust me this is the fix, and yes you can only make one 729 call at a time. Jason Price On 11/23/05, Aaron Clauson [EMAIL PROTECTED] wrote: Hi, I have a very strange Asterisk SIP call signalling problem that is proving extremely difficult to track down. The problem is that any SIP INVITE request that is coming into Asterisk over a satellite connection from a Linksys Router or PAP2 is getting a Not Acceptable Here (codec error) from Asterisk. I've done all the normal checks on the allowed codecs in sip.conf but to no avail. I've even gone as far as writing a basic SIP stack to authenticate and send the INVITE request to Asterisk with exactly the same SDP payload to let me brute force different options in the SDP request to try an narrow it down that way. The preplexing thing from that length exercise is that if exactly the same INVITE request comes in from my app across the same satellite connection to Asterisk it gets 200 Ok'ed but coming from the Linksys PAP2 or WRT54GP2 it gets 488 Codec Not Acceptable Here'ed. The first time this happened we went through all the usual checks and got nowhere and the person drifted off and it was put down to something speicifc to that set up/connection. But now it's cropped up again with a different person who also just happens to be on a satellite connection but from a different provider, although it is possible both providers use the same infrastructure. In both cases incoming calls to the Linksys devices worked correctly it's just the outgoing calls from the devices to Asterisk that are getting the rejection. In the second case we can't put it down to something to do with the connection because the person has a Vonage service working no problems across the same satellite link we are getting the rejection on. The SIP trace is below and I'm wondering if anybody has ever seen something similar. The only thing I can think of is that it's somehow a timing issue I can't see how it can be a codec issue since the exactly the same SDP payload will get OK'ed if coming from my app. Is the Asterisk SIP stack sensitive to the any timings in the INVITE request? It seems highly unlikely but I just can't think of anything else. INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f From: XXX sip:[EMAIL PROTECTED];tag=831f2cca367c3ddfo1 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=XXX,realm=asterisk,nonce=489bfe04,uri=sip:018X [EMAIL PROTECTED],al gorithm=MD5,response=22f566e03a225047469d73bec5ab640c Contact: XXX sip:[EMAIL PROTECTED]:5061 Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 424 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 418210 418210 IN IP4 192.168.1.248 s=- c=IN IP4 192.168.1.248 t=0 0 m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
RE: [Asterisk-Users] PAP2 and ringing issues
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Humberto Aicardi Sent: 01 November 2005 17:17 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PAP2 and ringing issues Hi, I currently have several PAP2-NA units configured to an Asterisk box, everything works fine except from the fact that after dialing a number I can hear ringing tones. When I connect to the same Asterisk box using XLite or EyeBeam I hear only one, any ideas on what may be wrong on the PAP units? Hi Humberto, We had this problem with calls being sent to a PRI. The two ringtones were due to both an RTP audio stream being generated from the PRI (this is the one we wanted) and also a SIP 180 ringing response being sent by the same Asterisk server. I'm not sure why both are getting sent, in 1.0.7 I'm pretty sure they weren't. The fix was simply to set progressinband=no in sip.conf on the Asterisk server with the PRI. The reason you only get the doble ring on one UA and not others seems to be entirely down to the UA. In our case the Linksys units act passed on both ringing indications where as Cisco IP Phones disregarded the SIP 180 and just passed on the RTP. Hth. Aaron ___ --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] Add Contexts Dynamically
Hi, Is it possible to dynamically add contexts to the dial plan in any way? Extensions can be added from the console and therefore also from MAPI but their doesn't appear to be anyway to add a new context apart from reloading the configuration files. The reason I ask is my dialplan is getting quite large and with about 100 changes a day I'm just getting nervous about continually reloading the whole thing everytime. Thanks, Aaron ___ --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] WRT54GP2 (WiFi + ATA)
Hi, If anyone has either: - Found a company which ships these units outside the US, - Got one of the units and tried to unlock it from Vonage. Please post. (The Linksys WRT54GP2 is the first acceptably priced unit that has a router, WiFi and an ATA, at least that I know of). Aaron __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Working Asterisk With Vonage
Hi, I haven't worked with Vonage myself but I usually get this error back from my termination provider when the number I have sent them is incorrect. It might be worth checking you have used the correct prefix (011 or 00) and area code etc. Regards, Aaron Hi ! I have been working on making my asterisk server work with Vonage services. I have been able to recieve calls on my asterisk machine but i couldnt call through that account to other people. Means if i call a zap channel and then dial 1 314 652 ... then i get an error like Executing Dial(Zap/3-1, SIP/dialled number@sphone.vopr.vonage.net:5061) in new stack -- Called dialled number@sphone.vopr.vonage.net:5061 -- Got SIP response 404 Not Found back from 216.115.25.198 -- SIP/sphone.vopr.vonage.net-ec6e is circuit-busy == Everyone is busy at this time -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (local, 192512100488, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' whether i dial any number ... i get the same response... and always ... Can anyone guess what might be the problem ? in sip .conf my settings are : register = username:password@sphone.vopr.vonage.net:5061 [sphone.vopr.vonage.net] type = peer fromuser = username secret = password host = asterisk machine ip:5070 fromdomain=sphone.vopr.vonage.net dtmfmode=rfc2833 nat = yes canreinvite=no In extensions.conf i have done : exten = _1.,1,Dial,SIP/[EMAIL PROTECTED]:5061,tr exten = _1.,2,Hangup Please help me in this reagard. Regards , Usman. === ___ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Running Asterisk on Linksys Router
Hi, I don't know if I missed something on the recent posts regarding running * on the linksys boxes (couldn't make any sense of the gifs that were posted??)? Getting back to the original question, does anyone know where the firmware or source for a linksys box running * can be obtained? Aaron ___ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZyXEL P2602HW (WiFi + ATA Router)
Hi, Has anyone had any luck getting one of the new ZyXEL P2602HW routers working with *?? These units look good on paper: DSL modem, 802.11g, 4 Port Ethernet, 2 x ATA plus all the bells and whistles in the firmware. It has 2 different SIP clients built in and I was able to get them registered to * and make outgoing calls but couldn't get them to answer calls (yes it's behind NAT and yes I have Grandstreams and softphones working fine behind the same NAT). I also noticed after making a call it started sending RTCP sender report packets back to * on what looked like an infinite loop. At this stage I turned it off. The price point is fairly high - more then a WiFi router + ATA - so I can't see it being too popular. Nevertheless these type of units could make life a lot easier for us VoIP provider folks that want to give customers a single box. Maybe version 2... Aaron ___ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] E100P and Colt Telecom (Europe)
Hi, Thanks a lot for the configs Fabe. I tried your zaptel.conf but I still get yellow and red alarms in zttool and * is unable to create any Zap channels (as expected with yellow and red alarms). I realise I will now have to start talking to Colt (in Ireland) to try and get the line up and running but if anyone has encountered this or something similar with Colt, or another provider in Europe, any tips would be greatly appreciated. Thanks, Aaron Message: 7 Date: Sat, 17 Jul 2004 10:38:26 +0200 From: Fabian Stelzer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E100P and Colt Telecom (Europe) Reply-To: [EMAIL PROTECTED] zaptel.conf span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=nl zone=de doesen't work correctly for me :( but nl does... zapata.conf switchtype=euroisdn pridialplan=unknown signalling=pri_cpe group = 1 channel = 1-15,17-31 context=incoming this is the base config that works with colt... the rest has to be configured to you needs... Regards Fabe __ Do you Yahoo!? Vote for the stars of Yahoo!'s next ad campaign! http://advision.webevents.yahoo.com/yahoo/votelifeengine/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P and Colt Telecom (Europe)
Hi, Has anyone connected * to a Colt E1 line in Europe? If so could you send me the zaptel.conf and zapata.conf. Thanks, Aaron __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Future WinCE IP Phone
[Kevin Walsh Wrote] Marvellous. Microsoft will bring their legendary stability, security and reliability to the VoIP world. Oops - there goes my lunch. Maybe but looking past that what the unit will bring is a programmable touch screen GUI on a hard VOIP phone. And being a Microsoft product it's going to have the familar look and feel, outlook synchronisation, office integration etc. etc. that 90% of the computer users on the planet know how to work. Aaron __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/IAX to PSTN setup time
Hi, I have started some users terminating calls from my asterisk server to the PSTN through a couple of termination providers. The biggest problem I am having is the time it takes to initially set the call up. It regularly exceeds twenty seconds. I can work around this with failing over to another provider or increasing the timeout but people are used to call setup times of 5 to 10 seconds. I imagine this is a fairly common situation. Does anyone know the reason for the large setup time and/or how to reduce it? Aaron __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Future WinCE IP Phone
Hi, Found a nice little video about a prototype phone from broadcom currently sitting in Microsoft WinCE lab. The video is at: http://channel9.msdn.com The video in question is an interview with Mike Hall titled Windows CE and Windows Embedded Lab Tour. The clip dealing with the VOIP phone is right at the start so you don't need to watch the whole thing (although there is some more interesting stuff such as a programmable sewing machine...). Couldn't find any info about the phone on the broadcom site. It will be nice when the phones are this smart (as well as an order of magnitude cheaper) and VOIP starts selling itself. Skype also might have an even tougher time when MSN messenger intergrates voice again; glad I didn't contribute to the 11 sterling million funding round. Aaron __ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: X100P in Switzerland
Hi, I had a similar problem for a while in Ireland. Eventually after much hair tearing I decided it must be something to do with the phone socket and commenced to make a direct conenction between the twisted pair and the X100P socket. Low and behold it worked. After more mucking around I found I could get the card to work, and get the red alarm removed, by jiggling the RJ11 cable in the phone socket. I would plug a analogue phone into the X100P and then a cable from the line in on the card to the phone socket. By moving the cable in and out of the socket I could get the signal passed through to the phone and at the same time clear the red alarm. I am pretty sure this has something to do with the line impedance but despite having a dim distant electronic engineering degree don't really understand it?? hth, Aaron quote Hi Does anybody if the X100P works in Switzerland? We can't get a line to PSTN. When I run zttool it shows me always a red alert. I can make and receive calls with an anlog phone plugged in the phone connector. I've compiled and configured the card according to the wiki. Everything seemed to be ok. Is there a way to debug this? Regards Reto /quote __ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cost of IP Phones, or Isn't It Just Software?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Sandee Sent: Wednesday, June 16, 2004 8:45 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cost of IP Phones, or Isn't It Just Software? Am I dreaming? Firstly one would have to wonder if Digium will be taking the next step to produce a handset based on their, yet to be released, IAXy. The IAXy appears to be a possible core just add the keypad and handset. Secondly IF a device could be built there MAY be business models that wouldn't need centralised sales and marketing budgets. For example many people on this list are running VOIP businesses where it MAY make sense to give CHEAP handsets away in order to gain subscribers and then recoup the costs from call or subscription fees. Thirdly what sort of expertise would the group be able to pull together? Software seems to be the main competency and hardware is a different kettle of fish. That being said these days a lot of consumer hardware is going the way of reference designs (broadcom and WiFi routers for example) with OEMs differentiating on the software or even just the user interface. IF a good hardware reference design was available for a VOIP handset software would POSSIBLY become the primary task and therefore POSSIBLY suit the expertise of this group... Regards, Aaron __ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] International Talking Clocks
Hi, Does anyone know of a list of internationally accessible PSTN talking clocks? I find talkjing clocks a good way to test the call quality to a particular country. There are a quite a few available in the US but the only other two countries I have found numbers for are the UK and Sweden. Other countries obviously have them but they generally don't seem accessible from international numbers. Talking Clock Numbers: Sweden: +46-3390510 UK: +44-8451249068 US: +1-2027621401 Anyone know (or provide access to) any others? Regards, Aaron __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GUI Design Ideas Request
Hi, Yes I am contemplating writing yet another GUI application for *. However I thought before I start coding away I would see if anybody had any good ideas about the interface. I have had a look around at the other * GUIs and also a quick search of other PABX GUIs but to my mind there was nothing that stuck out as inuitive, powerful or broad enough. Anyway if anyone has any ideas especially in the form of images I would be very interested in seeing them. I hacked together some ideas myself, I apologise in advance for my very poor grpahic design skills: http://www.blueface.ie/asterisk/AsteriskGUI.png There still seem to be numerous people mentioning the lack of a powerful GUI as an * shortcoming. If a good interface design was found, i.e. pictures with some workflow descriptions, it would certainly be encouraging to any prospective GUI coders. Regards, Aaron __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No sound for MusicOnHold and SayDigits
Hi, I am unable to get any music or sounds played with the MusicOnHold or SayDigits commands. I do get sound from the Playback and Background commands. I have gone through the process of installing mpg123 and putting the link in usr/bin (and usr/local/bin). For the MusicOnHold command I can see the call come into * and the command get executed I just get no sound on the phone. The * console messages are: --Executing MusicOnHold(SIP/phone1-6b05, ) in new stack -- Started music on hold, class 'default', on SIP/phone1-6b05 I am using a SIP Grandstream phone with no NAT between it and *. Excerpts from my configuration files are: musiconhold.conf: [classes] default=quietmp3:/var/lib/asterisk/mohmp3 zapata.conf: ... musiconhold=default ... extensions.conf: ... [sip] exten=9216,1,MusicOnHold() As far as I am aware these commands have nothing to do with whether or not there is a sound card present? Thanks, Aaron __ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: X100P Red Alarm Ireland
Thanks for the suggestion about checking the wiring of my telephone socket! I was able to get my X100P to pass through the signal and get rid of the Red Alarm in zttool, hallelujah!!! My understanding of the problem was that the X100P wants the POTS signal on pins 2 5 whereas the Irish sockets are wired up for pins 3 4. I do have another problem with the socket that was installed by the telco (Eircom). The only way I can get the X100P to accept the signal is by connecting the POTS pair directly to the RJ11 coming from the card. If I try and go through the socket no signal gets through. I checked the connections through the socket and I have the pins wired correctly so I can only assume that the in built resistance of the socket is not letting enough current through to the card??? The socket is manufactured in the UK and has ISDN writen on it as well as having an RJ45 connector. I would love to know what is going on here... Thanks, Aaron __ Do you Yahoo!? SBC Yahoo! - Internet access at a great low price. http://promo.yahoo.com/sbc/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] X100P Ireland Red Alarm (AR Tarzi)
Ahhh this could be my problem! I just checked which wires on the RJ11 cable had a voltage across them and it was the yellow and green (3 4?). From what someone posted the other day it's supposed to be Bumble Bee and Christmas Tree. I did have to get a technician out to fix my line when it was first installed because it was dead. Maybe he wired it up incorrectly or maybe they just do it different here in Ireland. I'll buy a crimping tool tomorrow and try out different combinations. Thx. Aaron From: AR Tarzi [EMAIL PROTECTED] Important. 1. Try the phone (set) directly on the line.. - confirm you have = dialtone 2. Make sure the phone is picking up the line from pins 3 4 on the = RJ11 ONLY .. i.e. if your line is using a non-standard interface (and so = does your phone) this is a possible failure - not of the card, but of = the connection. __ Do you Yahoo!? SBC Yahoo! - Internet access at a great low price. http://promo.yahoo.com/sbc/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Ireland Red Alarm
Hi, Has anyone got the X100P to work with an anlogue line in the Republic of Ireland? I have the X100P installed but zttool indicates a Red Alarm status on the card. It is on its own interrupt and I have tried different PCI slots but all to no avail. Are there any alternatives to the X100P that can work with asterisk and are likely to work in Ireland? Thanks, Aaron __ Do you Yahoo!? SBC Yahoo! - Internet access at a great low price. http://promo.yahoo.com/sbc/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: Re: [Asterisk-Users] X100P Ireland Red Alarm
Hi, I suspected that I the analogue phone should have got a pass through signal when the power was off to the server, unfortunately it doesn't. I kept asking digium support about that but they didn't give me an answer. The problem is how do I identify whether the X100P is incompatibel with the network or faulty without possibly wasting another USD100??? Aaron On Sat, 2004-05-15, Eric Wieling wrote: If you plug a regular ANALOF phone into the second port on the X100P do you get dialtone? The second port is hardwired to the first port, so if you don't get dialtone on the second port then the phone line you have plugged into the X100P is not working. On Sat, 2004-05-15 at 03:17, Aaron Clauson wrote: I have the X100P installed but zttool indicates a Red Alarm status on the card. It is on its own interrupt and I have tried different PCI slots but all to no avail. __ Do you Yahoo!? SBC Yahoo! - Internet access at a great low price. http://promo.yahoo.com/sbc/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
