Re: [asterisk-users] Symbian IAX client
Yeah, it would be very neat. NAT is such a pain, roll on IPv6 :) On Fri, 2007-02-16 at 20:46 +0800, Vernier Umali wrote: I also would like to know if there is an application like this. The most i've tried in a mobile device is using PPCIAX for the pocketpc. Any comments also on the feasibility of developing something like this if the application is not yet available. On 2/15/07, Peter Spikings [EMAIL PROTECTED] wrote: Hi all, Does anyone know of an IAX client for Symbian? I have an e61 and would like to make calls through my home Asterisk box from places where I have WiFi access, as NAT is in the way I suspect that it'll be a pain to get SIP working like that as the NAT router doesn't do SIP connection tracking. Thanks, Peter. This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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 ___ --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 This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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] Symbian IAX client
Hi all, Does anyone know of an IAX client for Symbian? I have an e61 and would like to make calls through my home Asterisk box from places where I have WiFi access, as NAT is in the way I suspect that it'll be a pain to get SIP working like that as the NAT router doesn't do SIP connection tracking. Thanks, Peter. This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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] Re: mISDN
Hi all, How come I occasionally get messages with the subject NOT containing [asterisk-users] ? It stops my filters working! Thanks, Peter Spikings. On Mon, 2007-02-05 at 10:06 +, Tomislav Parčina wrote: Hi list! How to make outgoing call thru other mISDN channel group of all channels on first group are busy? I believe I'll need to - Check of there is free channel on group1 - if there is free channel call thru group1 - if there are no free channels call thru group2 -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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: filtering [was: Re: [asterisk-users] Re: mISDN]
On Mon, 2007-02-05 at 12:31 +0200, Tzafrir Cohen wrote: On Mon, Feb 05, 2007 at 10:15:39AM +, Peter Spikings wrote: Hi all, How come I occasionally get messages with the subject NOT containing [asterisk-users] ? It stops my filters working! I don't know. However there are better ways to filter than by the subject line. Mailman injects standard mailing list headers to the list. Any decent mailer should be able to filter by those. Yes, that's very true. However it is still worth pointing out... Fortunetly you seem to use a decent mailer (Evolution) that supports filtering by mailing list headers, and don't even have to resort to filter by a custom mail message header. Yes, I'll set that up properly. Thanks, Peter. This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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] Size of realtime appdata field under MySQL
Hi all, I'm trying to use a bigger appdata column for realtime, the reason being that I'm moving to a new setup where the SIP devices are named according to the name of the user and some of my dial/page commands need to dial a goodly number of phones which then exceeds the 255 max size of the column. I've tried turning the column into a text and Asterisk copes with that but still truncates it somewhere. Is this possible / is there a constant I can change somewhere? :) Thanks, Peter. This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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] Asterisk RealTime Question, Please help
Hi, OK, that will enable the auto generation of a context but as the new context won't have a switch statement it doesn't help with this problem... I may try writing a default switch if no matching context found type patch. Peter. On Mon, 2006-03-13 at 20:51 +0200, Benchev wrote: I was able to install Asterisk and Asterisk-addons and use them successfully. But I have a problem now, I have many contexts and it looks like Asterisk is unable to find the context given directly in Mysql DB unless I specify it in Extensions.conf to switch it to RealTime. If I add a new context in Mysql then I have to add it in Extensions.conf and reload extensions whenever I need a new context. Please tell me if there is a way to avoid all this and make Asterisk take contexts directly from Mysql without mentioning that context in Extensions.conf. If this is possible then I can make my Asterisk RealTime actually and modify contexts directly in Mysql. Idea from the wiki: ; If regcontext is specified, Asterisk will dynamically create and destroy a ; NoOp priority 1 extension for a given peer who registers or unregisters with ; us. The actual extension is the 'regexten' parameter of the registering ; peer or its name if 'regexten' is not provided. More than one regexten may ; be supplied if they are separated by ''. Patterns may be used in regexten. ; ;regcontext=sipregistrations That means that you should creat a mother context in extensions.conf: [sipregistrations] But first I would try to add a field regcontext along with regexten(which already there) in sip_users table since for the trick to work ... read http://www.voip-info.org/wiki-Asterisk+sip+regcontext Hope this will give you a clue. Benchev ___ --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 This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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] Asterisk RealTime Question, Please help
Hi all, I was about to ask this question so here's an attempt to not let it get lost in the general noise on the list! Thanks, Peter Spikings. On Mon, 2006-03-13 at 01:52 -0600, [EMAIL PROTECTED] wrote: Hi All, I was able to install Asterisk and Asterisk-addons and use them successfully. But I have a problem now, I have many contexts and it looks like Asterisk is unable to find the context given directly in Mysql DB unless I specify it in Extensions.conf to switch it to RealTime. If I add a new context in Mysql then I have to add it in Extensions.conf and reload extensions whenever I need a new context. Please tell me if there is a way to avoid all this and make Asterisk take contexts directly from Mysql without mentioning that context in Extensions.conf. If this is possible then I can make my Asterisk RealTime actually and modify contexts directly in Mysql. Thanks for you help and time, Manoj. ___ --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 This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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] Call queuing question
Hi, Could I have clarification on the logic in app_queue which treats no answer as needing a retry? What I want to do is have all calls firstly always go to phone A, then if there is no answer make it call B or C in a round robin fashion. The obvious thing to do is put a penalty on B C but then if phone A doesn't pick up it just keeps retrying which isn't what I want as the person with phone A on their desk may be absent for a couple of minutes. Could I ask why no answer is treated as needing a retry rather than moving up to the next penalty group? Thanks, Peter Spikings. This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details. ___ --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
Re: [Asterisk-Users] Call queuing question
Hi, Using agents would involve the user having to remember to login again every time they leave their desk as it would only be useful if they were auto-logged off ;) I've tried playing with the timeouts and have found that the timeout parameter to queue causes it to return to the dialplan after that long (as if the n option was set, which it wasn't). The timeout parameter on the queue moves onto the next phone at the same penalty and never advances to the next penalty. It's the latter behaviour that I find puzzling, surely a timeout when the strategy is ringall or all other phones at the current penalty have also timed out should make it advance to the next penalty Cheers, Peter. On Wed, 2005-10-19 at 13:50 +0200, Lenz wrote: Hello, if you use a mechanism like agents, * will know that there is nobody at the first level of penalty and route the call to the other level. A different approach could be to have a queue ring A for say 20 second, timeout, route the call to a second queue where B and C are. This should fix yoiur problem. Bye l. On Wed, 19 Oct 2005 11:49:16 +0200, Peter Spikings [EMAIL PROTECTED] wrote: Hi, Could I have clarification on the logic in app_queue which treats no answer as needing a retry? What I want to do is have all calls firstly always go to phone A, then if there is no answer make it call B or C in a round robin fashion. The obvious thing to do is put a penalty on B C but then if phone A doesn't pick up it just keeps retrying which isn't what I want as the person with phone A on their desk may be absent for a couple of minutes. Could I ask why no answer is treated as needing a retry rather than moving up to the next penalty group? Thanks, Peter Spikings. This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details. ___ --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 This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details. ___ --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] Listening for DTMF when dialling
Hi all, I want to set up an extension which dials a group of phones while at the same time plays a message (Press 1 to leave a message) and listens for DTMF. I haven't played around yet but the way I read the docs this isn't possible as Thanks, Peter Spikings This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details. ___ --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] Listening for DTMF when dialling (sorry, accidentally sent the previous message too early!)
Hi all, I want to set up an extension which dials a group of phones while at the same time plays a message (Press 1 to leave a message) and listens for DTMF. I haven't played around yet but the way I read the docs this isn't possible as the dial command doesn't have appropriate options and takes complete control of the channel. However surely this is a normal thing to want to do? Am I right thinking it's not possible? Are there any plans to have (say) a fork command which splits the channel into 2 or more threads (passing audio from the first specified to the caller) and another command (and option to dial) which make * abandon the other threads and join the caller to the current thread? I'd say it would make things like this a lot easier and * even more flexible ;) Thanks, Peter Spikings This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details. ___ --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] Playing sounds while dialling
Hi all, Does anyone know a way of playing a sound while the dial command is running? I want to play a sound every 10 seconds while relevant phone(s) are ringing and have ring tones played to the caller in the gaps. I am aware of the fact that you can play a class of MoH during the dial but I can see two problems with that firstly that the caller won't hear the sound from the start and secondly that ring tones may not sound right after having been encoded to MP3. I've tried using Background but that waits until the sound has finished before the Dial command is executed. Thanks, Peter Spikings This message has been comprehensively scanned for viruses, please visit http://virus.e2e-filter.com/ for details. ___ 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] Problem with IAX trunking
Hi all,
I'm trying to get IAX2 trunking between two * boxes and am having
extreme difficulty :) What happens is when the sending * server (the one
initiating the call) receives the ACCEPT back from the receiving server
it immediately replies with INVAL. I've checked the code and it seems to
be not matching the accept packet with the relevant item in the iaxs
array due to the following condition in the match function evaluating as
false:
if ((cur-peercallno == callno) ||
((dcallno == cur-callno) !cur-peercallno))
dcallno and cur-callno are identical and =16384 indicating a trunk
call but cur-peercallno is non-zero hence it fails.
I'm using Asterisk 1.0.7 on both servers. From the code this looks like
a bug but the fact that Google searches show no-one having the same
problem suggests a config problem ;) I'll attach the configs and
ethereal traces of this happening.
Now it gets more interesting :) If I change type=friend to type=peer in
both config files the call goes through fine but upon closer inspection
it seems to trunk in the outgoing direction only. The streams coming
back when two calls are made seem to be in their own packets with low
call numbers (16384).
If anyone would like further information then just ask :) If it turns
out to be a bug then I'd be happy to attempt a fix.
Many thanks,
Peter Spikings
This message has been comprehensively scanned for viruses,
please visit http://virus.e2e-filter.com/ for details.
Frame 1 (110 bytes on wire, 110 bytes captured)
Arrival Time: May 13, 2005 09:54:04.252924000
Time delta from previous packet: 0.0 seconds
Time since reference or first frame: 0.0 seconds
Frame Number: 1
Packet Length: 110 bytes
Capture Length: 110 bytes
Protocols in frame: eth:ip:udp:iax2
Ethernet II, Src: 00:50:da:b2:3f:aa, Dst: 00:90:fb:08:2a:db
Destination: 00:90:fb:08:2a:db (Portwell_08:2a:db)
Source: 00:50:da:b2:3f:aa (3com_b2:3f:aa)
Type: IP (0x0800)
Internet Protocol, Src Addr: 10.10.8.253 (10.10.8.253), Dst Addr: 10.10.8.252
(10.10.8.252)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00)
0001 00.. = Differentiated Services Codepoint: Unknown (0x04)
..0. = ECN-Capable Transport (ECT): 0
...0 = ECN-CE: 0
Total Length: 96
Identification: 0x0014 (20)
Flags: 0x04 (Don't Fragment)
0... = Reserved bit: Not set
.1.. = Don't fragment: Set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 64
Protocol: UDP (0x11)
Header checksum: 0x145d (correct)
Source: 10.10.8.253 (10.10.8.253)
Destination: 10.10.8.252 (10.10.8.252)
User Datagram Protocol, Src Port: 4569 (4569), Dst Port: 4569 (4569)
Source port: 4569 (4569)
Destination port: 4569 (4569)
Length: 76
Checksum: 0x7a39 (correct)
Inter-Asterisk eXchange v2
Packet type: Full packet (1)
.100 = Source call: 16384
.000 = Destination call: 0
0... = Retransmission: False
Timestamp: 12
Outbound seq.no.: 0
Inbound seq.no.: 0
Type: IAX (6)
IAX type: NEW (1)
Information Element: Protocol version (0x0B)
IE id: Protocol version (0x0B)
Length: 2
Protocol version: 0x0002
Information Element: Number/extension being called (0x01)
IE id: Number/extension being called (0x01)
Length: 3
Number/extension being called: 601
Information Element: Calling number (0x02)
IE id: Calling number (0x02)
Length: 3
Calling number: 299
Information Element: Name of caller (0x04)
IE id: Name of caller (0x04)
Length: 14
Name of caller: Peter Spikings
Information Element: Desired language (0x0A)
IE id: Desired language (0x0A)
Length: 2
Desired language: en
Information Element: Desired codec format (0x09)
IE id: Desired codec format (0x09)
Length: 4
Desired codec format: GSM compression (0x0002)
Information Element: Actual codec capability (0x08)
IE id: Actual codec capability (0x08)
Length: 4
Actual codec capability: 0xf802
...0 = G.723.1
compression: Not supported
..1. = GSM compression:
Supported
.0.. = Raw mu-law data
(G.711): Not supported
0... = Raw A-law data
(G.711): Not supported
...0
