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: sip:82.70.154.145;ftag=as3606b893;lr=on 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 sip:[EMAIL PROTECTED];tag=as3606b893 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] 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: sip:82.70.154.145;ftag=as3606b893;lr=on list_route: hop: sip:[EMAIL PROTECTED] 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 sip:[EMAIL PROTECTED];tag=as3606b893 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED]:5061 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 sip:[EMAIL PROTECTED]:5061;tag=as60a4b224 To: sip:[EMAIL PROTECTED]:5061 Contact: sip:[EMAIL PROTECTED]:5061 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: sip:[EMAIL PROTECTED]:5061 From: unknown sip:[EMAIL PROTECTED]:5061;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: sip:[EMAIL PROTECTED]:5061;tag=d416591c6d2e2378i1 From: unknown sip:[EMAIL PROTECTED]:5061;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: sip:[EMAIL PROTECTED]:5061;tag=d416591c6d2e2378i1 From: unknown sip:[EMAIL PROTECTED]:5061;tag=as60a4b224 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.4.3:5061;branch=z9hG4bK7792da43 Contact: PSTN Line sip:[EMAIL PROTECTED]:5061 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
[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: sip:[EMAIL PROTECTED];user=phone;tag=2361964166 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;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: sip:[EMAIL PROTECTED];user=phone;tag=2361964166 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: sip:[EMAIL PROTECTED];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: sip:[EMAIL PROTECTED];user=phone;tag=2361964166 To: sip:[EMAIL PROTECTED];user=phone;tag=as0771f231 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;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] 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: sip:[EMAIL PROTECTED];user=phone;tag=2361964166 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;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: sip:[EMAIL PROTECTED];user=phone;tag=2361964166 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: sip:[EMAIL PROTECTED];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: sip:[EMAIL PROTECTED];user=phone;tag=2361964166 To: sip:[EMAIL PROTECTED];user=phone;tag=as0771f231 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;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
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
[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] 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
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
[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] 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/101SIP/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
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
[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
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] 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 ___ 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] 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 ___ 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
