Re: [Asterisk-Users] NAT and sip issues
The rtp audio is going phone to phone, not via asterisk. This is one of the reasons I am trying to set up SER with Asterisk. >> I have an asterisk server behind NAT - no audio on the test external >> calls >> I >> have tried making so far. >> >> Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No >> solution >> evident from there, sounds like I have case 9. I would have thought that >> all I >> would have to do is port foward and have the external IP on the asterisk >> server, >> which I have done >> >> I have fowared 5060UDP, 8000UDP, and 35000 to 37000 UDP to the internal >> IP >> (192.168.1.115) >> >> I have put 35000 and 37000 into the rtp.conf as the start/end ports >> >> extracts of sip.conf: >> >> externip = 60.234.129.154 >> localnet = 192.168.1.115 >> localmask = 255.255.255.0 >> >> >> [88] >> type=friend >> secret=** >> dtmfmode=rfc2833 >> nat=yes >> host=dynamic >> canreinvite=no >> >> >> Trying with xlite at the other end >> >> Registered ok, can dial both ways, just no audio at all. >> >> In the log of xlite (cant see it at the moment as im not vnc'd in at the >> moment) >> it showed the xlite machines private IP address on some of the >> transactions that >> were logged. >> >> The client has a dynamic IP address so cant really be specified anywhere >> in the >> xlite configuration, I am also not sure on all the different firewall >> types. >> >> I was under the impression that there was no need to configure any >> portfowards >> at the sip softphone end. >> >> I will hopefully be using xlite or similar from a location with a very >> locked >> down firewall environment. I want to check all works on a normal nat >> router >> before trying it behind the nasty nat/firewall at this location. >> ___ >> 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 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] ser and asterisk
Hello, I have noticed when ser forwards to asterisk, the last "registered" host from ser is always the subsequent callee whichever client dials. i.e. 4561 registers 4562 registers 4563 registers 4562 calls 4561. Asterisk shows 4563 dialing 4561. I am forwarding registrations and invites to asterisk. Is this correct? Many thanks, Spencer ___ 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] asterisk dials random number when receiving incoming call
Hello,
I have found asterisk is dialing a random number when it recieves a call,
would anyone know why? The first thing I noticed found peer 4563 (this is
a n Xlite Client)
Many thanks,
Spencer
SIP Debugging Enabled
spitfire*CLI>
<-- SIP read from 82.70.154.145:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Max-Forwards: 10
Record-Route:
Via: SIP/2.0/UDP 82.70.154.145;branch=z9hG4bK2922.6c170001.0
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
From: "unknown" ;tag=as3606b893
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: MSS VoIP Gateway
Date: Sat, 14 May 2005 01:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 31661 31661 IN IP4 213.166.5.129
s=session
c=IN IP4 213.166.5.129
t=0 0
m=audio 14474 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (15 headers 12 lines)---
Using latest request as basis request
Sending to 82.70.154.145 : 5060 (NAT)
Found peer '4563'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 213.166.5.129:14474
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 448715046363 in local-sip
list_route: hop:
list_route: hop:
Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK2922.6c170001.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:5060;branch=z9hG4bK6a346c4d
From: "unknown" ;tag=as3606b893
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
---
-- Executing Dial("SIP/4563-5e36",
"SIP/[EMAIL PROTECTED]:5061|60|r")
We're at 192.168.4.3 port 35002
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.4.5:5061:
INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
From: "unknown" ;tag=as60a4b224
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 14 May 2005 01:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 8318 8318 IN IP4 192.168.4.3
s=session
c=IN IP4 192.168.4.3
t=0 0
m=audio 35002 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called [EMAIL PROTECTED]:5061
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 100 Trying
To:
From: "unknown" ;tag=as60a4b224
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0
--- (8 headers 0 lines)---
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 180 Ringing
To: ;tag=d416591c6d2e2378i1
From: "unknown" ;tag=as60a4b224
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0
--- (8 headers 0 lines)---
spitfire*CLI>
<-- SIP read from 192.168.4.5:5061:
SIP/2.0 200 OK
To: ;tag=d416591c6d2e2378i1
From: "unknown" ;tag=as60a4b224
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43
Contact: PSTN Line
Server: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 233
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
ontent-Type: application/sdp
v=0
o=- 3069797 3069797 IN IP4 192.168.4.5
s=-
c=IN IP4 192.168.4.5
t=0 0
m=audio 16452 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (12 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.4.5:16452
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop:
set_destination: Parsing for
address/port to send to
set_destination: set destination to 192.168.4.5, port 5061
Transmitting (no NAT) to 192.168.4.5:5061:
Re: [Asterisk-Users] realtime sip show peers no nat
Hello Matthew, Thank you, yes, nat is on, unfortunately, the contact points to the private IP address behind 212.74.112.53, but at least now I have somehting else to work on. I have cc'd the mailing list because I think it would be useful for others. Many thanks for your help, Spencer > To correctly verify if NAT is on a peer or not: > > realtime load sippeers name 5561 (look for the NAT column, should be "yes" > or > "no") > > if you need to change: > > realtime update sippeers name 5561 nat yes (or nat no) > > then do: > > sip prune realtime 5561 > > then: > > "sip show peer 5561 load" > > It should correctly display your nat'd option now. > > -Matthew > > Quoting "G.Marshall" <[EMAIL PROTECTED]>: > >> >> Hello >> >> sip show peers does not mark hosts as NAT even though sip.conf and >> sip_peers table has nat=yes. >> >> spitfire*CLI> sip show peers >> Name/username HostDyn Nat ACL Mask >> Port Status >> voipuser.org/gdsm 216.127.66.119 N 255.255.255.255 >> 5060 Unmonitored >> 5560/5560 192.168.4.5 D N A 255.255.255.255 >> 5060 Unmonitored >> 5561/5561 192.168.4.5 D N A 255.255.255.255 >> 5061 Unmonitored >> 4561/4561 212.74.112.53D N 255.255.255.255 >> 8413 Unmonitored >> 4 sip peers [4 online , 0 offline] >> spitfire*CLI> >> >> asterisk listens on 192.168.4.3 and 82.70.154.145. The host >> 212.74.112.53 >> is the external (NAT) address for a sip phone whose LAN address is >> 10.44.16.163. >> >> sip debug shows the following >> spitfire*CLI> >> <-- SIP read from 212.74.112.53:8413: >> REGISTER sip:82.70.154.145 SIP/2.0 >> Via: SIP/2.0/UDP 10.44.16.163:5060 >> From: ;tag=2361964166 >> To: >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 REGISTER >> Contact: >> ;expires=120 >> User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) >> Content-Length: 0 >> >> >> --- (9 headers 0 lines)--- >> Using latest request as basis request >> Sending to 10.44.16.163 : 5060 (NAT) >> Transmitting (NAT) to 212.74.112.53:8413: >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.44.16.163:5060;received=212.74.112.53;rport=8413 >> From: ;tag=2361964166 >> To: >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER >> Expires: 120 >> Contact: ;expires=120 >> Content-Length: 0 >> >> >> --- >> Transmitting (NAT) to 212.74.112.53:8413: >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 10.44.16.163:5060;received=212.74.112.53;rport=8413 >> From: ;tag=2361964166 >> To: ;tag=as0771f231 >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER >> Expires: 120 >> Contact: >> ;expires=120 >> Date: Fri, 13 May 2005 01:59:09 GMT >> Content-Length: 0 >> >> >> --- >> Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms >> >> >> Does anyone know how to rectify this? By the looks of things, >> >> Many thanks, >> >> Spencer >> >> ___ >> 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 >> > > > > > This message was sent using IMP, the Internet Messaging Program. > > ___ 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] realtime sip show peers no nat
Hello sip show peers does not mark hosts as NAT even though sip.conf and sip_peers table has nat=yes. spitfire*CLI> sip show peers Name/username HostDyn Nat ACL Mask Port Status voipuser.org/gdsm 216.127.66.119 N 255.255.255.255 5060 Unmonitored 5560/5560 192.168.4.5 D N A 255.255.255.255 5060 Unmonitored 5561/5561 192.168.4.5 D N A 255.255.255.255 5061 Unmonitored 4561/4561 212.74.112.53D N 255.255.255.255 8413 Unmonitored 4 sip peers [4 online , 0 offline] spitfire*CLI> asterisk listens on 192.168.4.3 and 82.70.154.145. The host 212.74.112.53 is the external (NAT) address for a sip phone whose LAN address is 10.44.16.163. sip debug shows the following spitfire*CLI> <-- SIP read from 212.74.112.53:8413: REGISTER sip:82.70.154.145 SIP/2.0 Via: SIP/2.0/UDP 10.44.16.163:5060 From: ;tag=2361964166 To: Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: ;expires=120 User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) Content-Length: 0 --- (9 headers 0 lines)--- Using latest request as basis request Sending to 10.44.16.163 : 5060 (NAT) Transmitting (NAT) to 212.74.112.53:8413: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.44.16.163:5060;received=212.74.112.53;rport=8413 From: ;tag=2361964166 To: Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: ;expires=120 Content-Length: 0 --- Transmitting (NAT) to 212.74.112.53:8413: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.44.16.163:5060;received=212.74.112.53;rport=8413 From: ;tag=2361964166 To: ;tag=as0771f231 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: ;expires=120 Date: Fri, 13 May 2005 01:59:09 GMT Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Does anyone know how to rectify this? By the looks of things, Many thanks, Spencer ___ 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
RE: [Asterisk-Users] Web interface Suggestions
> I think you will find AMP is about to implement a multi tenant solution. But does AMP deal with realtime? or just flat files the data for which is held in a db? >> >> Open Source project I assume. I am interested in this project do you Only open source. >> have a webpage about it? You can find the current version at https://www.dalmany.co.uk/asterisk/index.html I am open to suggestions and requests. Pages waiting incorporation include voicemail, sip users and sip peers. This only deals with Realtime, it does not replicate AMP with a db and flatfiles. It does not modify any flatfiles, only the realtime database so one has to know about realtime and how it works to get the full benefit. I am in the throws of moving house which is preventing me from developing it as quickly as I would like. >> >> Thanks, >> _ >> /-\ ndrew >> >> On 4/28/05, G.Marshall <[EMAIL PROTECTED]> wrote: >> > > Has anyone come across any software that can control >> adding/editing >> > > SIP extension properties and perhaps dial plan properties on a > context >> > > basis. What I mean is I would like it so an admin user from > Company A >> > > can manipulate >> > > properties for extensions in his context but not in another > Companies. >> I >> > > know AMP does something similar >> > > to this but from what I understand it does not allow for different >> users >> > > at different companies to control >> > > only things that pertain to them. >> > In my spare time, I am developing a php webfrontend to realtime > asterisk >> > database which modifies dialplan, users etc. Should not be too >> difficult >> > to add a login facility which means the user can see their own > context >> > only. >> > >> > Regards, >> > >> > Spencer >> > --- >> > https://www.dalmany.co.uk/dundi/dundi.php >> > https://www.dalmany.co.uk/asterisk/index.php > > ___ > 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 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
Re: [Asterisk-Users] Web interface Suggestions
> Has anyone come across any software that can control adding/editing > SIP extension properties and perhaps dial plan properties on a context > basis. What I mean is I would like it so an admin user from Company A > can manipulate > properties for extensions in his context but not in another Companies. I > know AMP does something similar > to this but from what I understand it does not allow for different users > at different companies to control > only things that pertain to them. In my spare time, I am developing a php webfrontend to realtime asterisk database which modifies dialplan, users etc. Should not be too difficult to add a login facility which means the user can see their own context only. Regards, Spencer --- https://www.dalmany.co.uk/dundi/dundi.php https://www.dalmany.co.uk/asterisk/index.php ___ 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] ser rtpproxy asterisk problems....
Hello, I have had asterisk running nicely for months. Now I have a need to intergrate with ser and rtpproxy. I have read all the literature I can find, but keep having the same problem. rtp proxying. sip (public) --> ser:5060 (public and private) --> asterisk:5061 and rtp (public and private) --> sip (private) I used nathelper.cfg with a minor modification to include rewritehostip to forward to asterisk. The private sip phone rings, but the rtp is not however proxied. Has anyone got a ser.cfg which they are willing to e-mail me to assit me in this problem? I am getting a sore head from banging it against the wall. Many thanks, Spencer --- https://www.dalmany.co.uk/dundi/dundi.php https://www.dalmany.co.uk/asterisk/index.php ___ 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
Re: [Asterisk-Users] Realtime with PostGreSQL ?
> Hi, > > Can I use ARA (Asterisk Realtime Architecture) with PostGreSQL database? Yes, I do, though I use odbc to connect to the postgres database Regards, Spencer --- https://www.dalmany.co.uk/dundi/dundi.php https://www.dalmany.co.uk/asterisk/index.php > > Regards, > > Fred OGUER > > ___ > 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 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] Rejected connect attempt
Hello, I have seen the following in my log files. For the life of me I can not work out why. Apr 22 22:10:40 NOTICE[19236] chan_iax2.c: Rejected connect attempt from 65.39.205.121, who was trying to reach 'i@' Would someone explain why, or point me in the direction I can read about it? Many thanks, Spencer --- https://www.dalmany.co.uk/dundi/dundi.php https://www.dalmany.co.uk/asterisk/index.php ___ 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
Re: [Asterisk-Users] Ring two extensions at the same time
> > On Thu, 14 Apr 2005 07:19:50 -0700 > Sean Kennedy <[EMAIL PROTECTED]> wrote: >> G.Marshall wrote: >> >>>Hello, >>> >>>I can not find anything on this, so it may not be >>>possible. >>> >>>I would like to dial one number which then rings at least >>>two extensions >>>at the same time. Not a hunt group, but ringing at the >>>same time as if >>>they were plugged into the same physical port. >>> >>>Does anyone know if this can be done, and if so how? >>> >>>Many thanks, >>> >>>Spencer >>> >> I know you can do Dial(SIP/101&SIP/102) and the like, >>but you specify you do not want this ( not a hunt group >>). How do you want the call to be handled when someone >>picks up a phone that's ringing? >> >> Sean > > Actually, that is what the "&" is for. It rings all those > phones at the same time and not in a hunt group. Using it > myself in a dialplan now to ring a zap channel, a sip > phone and an outside cell phone. All ring simutaneously > and when one phone is answered, all the others quit > ringing. > Thank you to both of you. I have used the "&" else where, but that is where one or other phone is live i.e. SIP phone or IAX phone but not both at the same time. I will give it a go. Thank you. ___ 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] Ring two extensions at the same time
Hello, I can not find anything on this, so it may not be possible. I would like to dial one number which then rings at least two extensions at the same time. Not a hunt group, but ringing at the same time as if they were plugged into the same physical port. Does anyone know if this can be done, and if so how? Many thanks, Spencer ___ 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
Re: [Asterisk-Users] RealTime
If you either e-mail me your current .conf files, or post your .conf files on the web somewhere I will take a look. I hope to release a web interface to the realtime database on Sunday at 1900BST. I will test your .conf files with a scratch database, and then come back to you. Would you also supply a desc or \d of the tables you have setup in the database. Spencer > Is there any better docs or step by steps other than what's in the Wiki for > Realtime setup? > > We have been trying to get this running and it's driving us batty.. > > It seems that the switch command is totally being ignored as far as we can tell. > > We are basically just getting an error telling us that the extension within > default can't be found. We have the extensions in the table and have the switch command pointing out to RealTime. > > If we put the extension in the text file it works, if we take it out of the > text file it breaks. > > We have searched and troubleshot all day, any other handy docs or step by steps out there? > > We are using the latest * via CVS.. > > Thoughts? > > Thanks.. > > > ___ > 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 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
RE: [Asterisk-Users] Web interface for realtime Mysql friends/peer
Hello, It was written to manage asterisk in a postgres database, not MySQL. It was written to add sip_users, sip_peers, dialplans etc. If you are still interested, I will send you the php. As I have written, it is for postgres, not MySQL. Spencer > Marshall, > > I am interested in seeing what you wrote to manage MySQL database > objects. > > By the way, latest version of OpenOffice comes with a MySQL > Administrator GUI to manage tables and data. This is something to look > at too. > > Seshu Kanuri > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of G.Marshall > Sent: Wednesday, April 06, 2005 2:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Web interface for realtime Mysql > friends/peer > >> Thanks >> >> But I was looking for a more complete solution like areski or astcc > I found nothing so I wrote my own, but they are for postgres. They are > not complete by no means. If you are interested, I will let you have a > look at what I have done, and if you provide constructive critisism, I > will be happy to release the php under the same licence as Asterisk. > >> >> Laurent >> At 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote: >> >>>phpmyadmin :) >>> >>>Matteo. >>> >>>Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha >>>scritto: >>> > Hello list, >>> > >>> > Does anyone know about a web/php interface to deal with users in >>> Realtime's >>> > Mysql database (sipusers and sippeers tables) ? >>> > >>> > Thanks in advance >>> > > >>> > Laurent >>> > >>> > >>> >>>___ >>>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 >>> >>> >>> >>>-- >>>No virus found in this incoming message. >>>Checked by AVG Anti-Virus. >>>Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 >> >> >> -- >> No virus found in this outgoing message. >> Checked by AVG Anti-Virus. >> Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 >> >> >> ___ >> 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 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 > > > NOTICE: If received in error, please destroy and notify sender. Sender > does not waive confidentiality or privilege, and use is prohibited. > > ___ 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
RE: [Asterisk-Users] ser <-> asterisk configs anyone?
> This may help, I just happen to be a google searching master :) I bow to the master ;) seriously though, thank you. > > http://www.voip-info.org/wiki-Asterisk+at+large I thought I had seen something somewhere. Plus I miss quoted voip-user.org for voip-info.org Thank you > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of G.Marshall > Sent: Wednesday, April 06, 2005 11:06 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] ser <-> asterisk configs anyone? > > > > I have searched high and low for these, but to no avail, nothing useful > back from google, nothing I could find on this mailing list, or > voip-user.org. > > Does anyone have any good urls and or pointers which will assist in > configuring SIP Express Router and Asterisk talking to each other on the > same machine? > > Many thanks, > > Spencer > > ___ > 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 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 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
Re: [Asterisk-Users] Web interface for realtime Mysql friends/peer
> Thanks > > But I was looking for a more complete solution like areski or astcc I found nothing so I wrote my own, but they are for postgres. They are not complete by no means. If you are interested, I will let you have a look at what I have done, and if you provide constructive critisism, I will be happy to release the php under the same licence as Asterisk. > > Laurent > At 11:12 06/04/2005 +0200, Matteo Brancaleoni wrote: > >>phpmyadmin :) >> >>Matteo. >> >>Il giorno mer, 06-04-2005 alle 20:05 +1100, Laurent Foulonneau ha >>scritto: >> > Hello list, >> > >> > Does anyone know about a web/php interface to deal with users in >> Realtime's >> > Mysql database (sipusers and sippeers tables) ? >> > >> > Thanks in advance >> > >> > Laurent >> > >> > >> >>___ >>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 >> >> >> >>-- >>No virus found in this incoming message. >>Checked by AVG Anti-Virus. >>Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 > > > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.308 / Virus Database: 266.9.3 - Release Date: 05/04/2005 > > > ___ > 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 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] ser <-> asterisk configs anyone?
I have searched high and low for these, but to no avail, nothing useful back from google, nothing I could find on this mailing list, or voip-user.org. Does anyone have any good urls and or pointers which will assist in configuring SIP Express Router and Asterisk talking to each other on the same machine? Many thanks, Spencer ___ 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] register => with realtime
Hello, I have realtime set up and working well. However, I can not work out how to do register => [EMAIL PROTECTED]/123 if there are no sip.conf, iax.conf etc. Any help with this would be much appreciated. Many thanks, Spencer ___ 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
Re: [Asterisk-Users] voicemail problems with CVS-HEAD
Thank you Mark, this has fixed my problem.
Many thanks,
Spencer
> All of my sounds are under /var/lib/asterisk/sounds. I don't have a
> directory /usr/share/asterisk. None of my configuration files have a
> pointer to a sounds directory so I'm assuming it's looking in
> /var/lib/asterisk/sounds by default.
>
> MARK.
>
> G.Marshall wrote:
>
>>Hello,
>>
>>I have moved from Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k (debian pkg) to
>>CVS-HEAD, and realtime. Compiled no problem and now running, with
>>realtime extensions and sip users in postgres (ODBC connection) database,
>>trunking also works.
>>
>>I have looked on google, wiki, and this mailing list, along with talking
>>to some peers, but to no avail.
>>
>>My problem revolves around voicemail. I have looked at the code, and
>>added a couple of ast_log entries to help debug. Those are the only
>>changes to the code. The voicemail.conf is exactly the same as it was
>> for
>>a working 1.0.7 version (I tried realtime voicemail but got the same
>>problem) I realise it can not find the voicemail playback files, which
>>are installed on the system (I have checked). I just do not understand
>>why not. Any help solving this would be much appreciated.
>>
>>Here is the console output.
>>
>> == No one is available to answer at this time (1:0/0/0)
>>-- Executing Voicemail("SIP/4560-4e18", "u4560")
>>Mar 25 04:46:23 WARNING[28472]: app_voicemail.c:1713 invent_message: fn
>>[/var/spool/asterisk/voicemail/default/4560/greet]
>>
>>Mar 25 04:46:23 WARNING[28472]: file.c:480 ast_openstream_full: File
>>en/vm-theperson
>>Mar 25 04:46:23 WARNING[28472]: file.c:485 ast_openstream_full: File
>>vm-theperson
>>Mar 25 04:46:23 WARNING[28472]: file.c:489 ast_openstream_full: File
>>vm-theperson does not exist in any format
>>Mar 25 04:46:23 WARNING[28472]: file.c:490 ast_openstream_full: File
>>vm-theperson does not exist in any format
>>Mar 25 04:46:23 WARNING[28472]: file.c:795 ast_streamfile: Unable to open
>>vm-theperson (format ulaw): No such file or directory
>>Mar 25 04:46:23 WARNING[28472]: app_voicemail.c:1730 invent_message:
>>ast_streamfile [vm-theperson]
>>
>> == Spawn extension (local-sip, 4560, 2) exited non-zero on
>> 'SIP/4560-4e18'
>>
>>cat /etc/asterisk/voicemail.conf
>>directoryintro=/usr/share/asterisk/sounds ; removing this makes no
>>; difference
>>dbuser=asterisk ; not sure this is needed
>>dbhost=localhost
>>dbname=asterisk
>>serveremail=asterisk
>>attach=yes
>>skipms=3000
>>maxsilence=10
>>silencethreshold=128
>>maxlogins=3
>>
>>
>>[zonemessages]
>>eastern=America/New_York|'vm-received' Q 'digits/at' IMp
>>central=America/Chicago|'vm-received' Q 'digits/at' IMp
>>central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred'
>> M
>>'hours'
>>
>>[default]
>>4560 => 4560,4560 Mailbox
>>
>>; end cat
>>
>>ls -l /usr/share/asterisk/sounds | grep vm-theperson
>>-rw-r--r-- 1 root root 2508 Mar 21 11:22 vm-theperson.gsm
>>
>>Any help in solving this would be much appreciated,
>>
>>Spencer
>>
>>___
>>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 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
>
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[Asterisk-Users] voicemail problems with CVS-HEAD
Hello,
I have moved from Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k (debian pkg) to
CVS-HEAD, and realtime. Compiled no problem and now running, with
realtime extensions and sip users in postgres (ODBC connection) database,
trunking also works.
I have looked on google, wiki, and this mailing list, along with talking
to some peers, but to no avail.
My problem revolves around voicemail. I have looked at the code, and
added a couple of ast_log entries to help debug. Those are the only
changes to the code. The voicemail.conf is exactly the same as it was for
a working 1.0.7 version (I tried realtime voicemail but got the same
problem) I realise it can not find the voicemail playback files, which
are installed on the system (I have checked). I just do not understand
why not. Any help solving this would be much appreciated.
Here is the console output.
== No one is available to answer at this time (1:0/0/0)
-- Executing Voicemail("SIP/4560-4e18", "u4560")
Mar 25 04:46:23 WARNING[28472]: app_voicemail.c:1713 invent_message: fn
[/var/spool/asterisk/voicemail/default/4560/greet]
Mar 25 04:46:23 WARNING[28472]: file.c:480 ast_openstream_full: File
en/vm-theperson
Mar 25 04:46:23 WARNING[28472]: file.c:485 ast_openstream_full: File
vm-theperson
Mar 25 04:46:23 WARNING[28472]: file.c:489 ast_openstream_full: File
vm-theperson does not exist in any format
Mar 25 04:46:23 WARNING[28472]: file.c:490 ast_openstream_full: File
vm-theperson does not exist in any format
Mar 25 04:46:23 WARNING[28472]: file.c:795 ast_streamfile: Unable to open
vm-theperson (format ulaw): No such file or directory
Mar 25 04:46:23 WARNING[28472]: app_voicemail.c:1730 invent_message:
ast_streamfile [vm-theperson]
== Spawn extension (local-sip, 4560, 2) exited non-zero on 'SIP/4560-4e18'
cat /etc/asterisk/voicemail.conf
directoryintro=/usr/share/asterisk/sounds ; removing this makes no
; difference
dbuser=asterisk ; not sure this is needed
dbhost=localhost
dbname=asterisk
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M
'hours'
[default]
4560 => 4560,4560 Mailbox
; end cat
ls -l /usr/share/asterisk/sounds | grep vm-theperson
-rw-r--r-- 1 root root 2508 Mar 21 11:22 vm-theperson.gsm
Any help in solving this would be much appreciated,
Spencer
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