Re: [Asterisk-Users] Hangup after a Timeout
What about playing a warning beep or IVR one minute before the call hangs up, can this be done? Thanks, Lei On Sat, Aug 02, 2003 at 09:39:38PM -0500, Martin Pycko wrote: Typically you use AbsoluteTimeout app. Martin On Sat, 2 Aug 2003 [EMAIL PROTECTED] wrote: hi everybody, can anybody pls tell me a way to hangup an answered Zap/SIP/IAX call, after a specified time period expires, like after 10, 15 minutes. Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Budgettone 100 102
-Original Message- From: Brian West [SMTP:[EMAIL PROTECTED] Sent: Sunday, August 03, 2003 12:45 AM To: [EMAIL PROTECTED] Subject:Re: [Asterisk-Users] Grandstream Budgettone 100 102 Everyone does now.. I don't get it.. they have a product we want.. but they wont or can't sell it. Guess they can't keep up with demand right now. bkw On Sat, 2 Aug 2003, Steven Honson wrote: I get a username/password prompt when I go to that page... This is what Brian West at Thu, Jul 31, 2003 at 12:03:53AM -0500 wrote: http://store.yahoo.com/grandstream-networks-inc/products.html I think that will clear it up. On Wed, 30 Jul 2003, Ricardo Villa wrote: I was quoted $75 and $85 USD today. Ricardo Villa http://www.telesip.net - Original Message - From: Joe Cooke [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 6:31 PM Subject: Re: [Asterisk-Users] Grandstream Budgettone 100 102 I was quoted the $75 and $85 USD prices from Grandstream direct about 2 months ago. I'm not sure if it makes a difference, but I live in the US. - Joe - Original Message - From: marrandy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:17 PM Subject: [Asterisk-Users] Grandstream Budgettone 100 102 Checking the earlier mails, it stated that the phones were $75 (100) $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 $85 rate ??? Regards...Martin -- Too much is just enough. -- Mark Twain, on whiskey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mysql CDR
hello all, I am using the msql cdr module to store cdr in db, I realised thatit does't capture the start and end time af a particular call record. Therefore I dive into the source code toadd the start and end timeinto the query (add something like cdr-start, cdr-end), but end up getting segfault. the original version of cdr_mysql.so works fine but Ineed the start time and end time of calling as well. I wonder what would Iget with cdr-start?the start time field in my dbis of type date or should i use varchar? thanks Foong
Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN
- Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 03, 2003 5:52 PM Subject: Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN Are you experiencing it over PRI ? Can you send the pri debug span spanno trace ?Is your asterisk/libpri code very recent ? I'm experiencing this both over a PRI line (E1), with july CVS, and over a normal ISDN BRI line, with latest CVS sources (taken about a week ago). I'v tried to debug both SIP and using messages (/var/log/asterisk/messages) but i found no useful informations. It's quite important to solve this problem 'cause i'm not able to call some *very* important number used for my job (Telecom HelpDesk, and so on). Thanks, -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP!!!!
[EMAIL PROTECTED] wrote: Hi When i make a call using oh323 is posible to make the ringing sound, to indicate the progress of the call As long as you don't Answer() the incoming H.323 call in extensions.conf you will be able to get the ringback tone from the phone. At this time when i make a call, i can't hear anything unless time out or the call is made Please hellp!! thanks Sorry for my english Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieving dialed number when overlap dialing?
On Sat, 2003-08-02 at 19:29, Mark Spencer wrote: How would you go about doing something like this? Add yet another variable to app_dial to hold the last called party number? We could make app_dial return with 0 given a specific option. Mark, doing it this way you can retreive the extension at which dial was called. The dial application will eat all digits received while 'overlap dialing', though. Say you have a dialplan: exten = _XXX.,1,Dial,Zap/g8/BYEXTENSION|20|HR exten = _XXX.,2,Dial,AGI(pm-store-number.agi) A user dialing 5551212 will be connected, but only 555 will be seen as the dialed extension by pm-store-number.agi. Thilo -- [netzquadrat] GmbH | Thilo Salmon Ronsdorfer Str. 74 | Fon: +49 211 302033 12 40233 Duesseldorf| Fax: +49 211 302033 22 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP clients not sending audio
Hi, I've got two SIP clients, one is X-Lite on NT, the other is KPhone on Linux and when I try either the echo test or voicemail demos, they fail to send any audio. They are both set up as type of friend in sip.conf so that they can send and receive calls. Using an IAX client like Gnophone, I have no problems. The troubling thing is that I'm almost certain that this was working prior to the latest CVS version. I've also played around with the config files so it is just as likely going to be my fault. I've run tcpdump and can see the call establishment, and the voice data coming from * to the SIP client, but then there is nothing from the SIP client back to *, which seems a little strange. Can anyone suggest anything, or at least a direction in which to look? Adam -- Adam Donnison email: [EMAIL PROTECTED] Saki Computer Services Pty. Ltd. 93 Kallista-Emerald Roadphone: +61 3 9752 1512 THE PATCH VIC 3792AUSTRALIAfax: +61 3 9752 1098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN
I've also got this problem over ISDN BRI using i4l. On Monday, August 4, 2003, at 09:17 AM, Stefano Finetti wrote: - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 03, 2003 5:52 PM Subject: Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN Are you experiencing it over PRI ? Can you send the pri debug span spanno trace ?Is your asterisk/libpri code very recent ? I'm experiencing this both over a PRI line (E1), with july CVS, and over a normal ISDN BRI line, with latest CVS sources (taken about a week ago). I'v tried to debug both SIP and using messages (/var/log/asterisk/messages) but i found no useful informations. It's quite important to solve this problem 'cause i'm not able to call some *very* important number used for my job (Telecom HelpDesk, and so on). Thanks, -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Paul Cheng Mátyás király ut 10 H-1121 Budapest HUNGARY [EMAIL PROTECTED] mobile: +36 30 381-9311 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] segmentation fault with asterisk and OH323
It's OK. I change my oh323.conf file and I don't have segmantation fault anymore. Thanks Rattana - Original Message - From: Rattana BIV [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 3:42 PM Subject: Re: [Asterisk-Users] segmentation fault with asterisk and OH323 By my phone I call H323 client (Netmeeting) I can talk. Everything is OK. But when I hangup I have the segmentation fault. These are the last asterisk log before the segmentation fault: -- Called 192.168.1.200 -- H323:193 answered CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2145 (ast_channel_bridge): Didn't get a frame from channel: CAPI[contr1/26]/1 DEBUG[23569]: File channel.c, Line 2213 (ast_channel_bridge): Bridge stops bridging channels CAPI[contr1/26]/1 and H323:193 Something strange is, when I call the phone from netmeeting it's work perfectly. What do you think ? (you can see my oh323.conf in attach file) Rattana - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 3:07 PM Subject: Re: [Asterisk-Users] segmentation fault with asterisk and OH323 Rattana BIV wrote: Hi, I got a segmantation fault When I call to computer (h323) from phone. I use asterisk-oh323 0.5.4 and chan_capi.0.2.2 drivers. More info (config files, screen log, backtrace of core file) is needed. Someone know where the problem ? Regards Rattana Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] small fix in chan_mgcp.c
there was a tiny memory leak in chan_mgcp.c
there's still lots of memory leaks AFACS, and I'm tracking on.
Still - I can't find the memory allocation routine allocating 2MB each time I
lift the handset of my mgcp phone ...
roy
diff -u -r1.16 chan_mgcp.c
--- channels/chan_mgcp.c2 Aug 2003 05:46:32 - 1.16
+++ channels/chan_mgcp.c4 Aug 2003 11:30:34 -
@@ -1261,7 +1261,10 @@
sdpLineNum_iterator_init(iterator);
while ((a = get_sdp_iterate(iterator, req, a))[0] != '\0') {
char* mimeSubtype = strdup(a); // ensures we have enough space
- if (sscanf(a, rtpmap: %u %[^/]/, codec, mimeSubtype) != 2)
continue;
+ if (sscanf(a, rtpmap: %u %[^/]/, codec, mimeSubtype) != 2) {
+ free(mimeSubtype);
+ continue;
+ }
// Note: should really look at the 'freq' and '#chans' params too
ast_rtp_set_rtpmap_type(sub-rtp, codec, audio, mimeSubtype);
free(mimeSubtype);
--
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 2254 5070 (work)
+47 9801 3356 (mobile)
Computers are like air conditioners.
They stop working when you open Windows.
___
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[Asterisk-Users] Any pointers for setting up PRI for incoming and outgoing calls?
Looking to setup a 6 channel PRI that is used for both incoming and outgoing calls. Incoming are presented 1-6 and outgoing 6-1. Anyone have a quick basic example? --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mem leak in logger.c?
hi trying to learn the * code a little better ... is it only me, or is the 'struct msglist *m' never freed? seems like it's only padded with more data along ... roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 2254 5070 (work) +47 9801 3356 (mobile) Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR
Hi, In the file Master.csv (var/log/asterisk/cdr-csv) we have stats of call. But Call from H323 client doesn't here. What sould have do in order to have this. Regards Rattana
Re: [Asterisk-Users] memory leak?
You can turn on Asterisk's internal malloc debugging in the Makefile. That can help track down the problem. Haven't found anything with that. from show memory summary: 200203 bytes allocated 1444 units total and after lifting the MGCP handset 200213 bytes allocated 1444 units total It'd be pretty wierd if this'd be enough to make the DRS value from ps axfv increase with 2048, though roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR
Rattana BIV wrote: Hi, In the file Master.csv (var/log/asterisk/cdr-csv) we have stats of call. But Call from H323 client doesn't here. What sould have do in order to have this. In oh323.conf: [general] amaFlags=billing Regards Rattana Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 CallerID
Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana
RE: [Asterisk-Users] Mysql CDR
Chee Foong, Here isa cdr_mysql2.c I rewrote. It captures all the billing information you will need. You will need to change the Makefile to compile it. Also attached is a script to create the database. Erik -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Chee FoongSent: Monday, August 04, 2003 2:24 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Mysql CDR hello all, I am using the msql cdr module to store cdr in db, I realised thatit does't capture the start and end time af a particular call record. Therefore I dive into the source code toadd the start and end timeinto the query (add something like cdr-start, cdr-end), but end up getting segfault. the original version of cdr_mysql.so works fine but Ineed the start time and end time of calling as well. I wonder what would Iget with cdr-start?the start time field in my dbis of type date or should i use varchar? thanks Foong cdr_mysql2.c Description: Binary data billing.sql Description: Binary data
[Asterisk-Users] limiting out going calls to a maximum duration
I want to limit my sons phone useage, by setting a 30min limit on out going calls from his room is there a simple way of doing this with asterisk? Thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some questions about a potential usage scenario for asterisk
Hi all, I'm new to the list and also new to the field of CTI. I'm looking for a solution to my problem (detailed in a mo) and after some extensive googling, came across asterisk but I'm not 100% sure if it will fulfill my needs - simply as I know nothing about CTI at this point. Ok my problem and desired solution goes like this. I have an intranet http-based application (developed in-house) which stores data in an RDBMS of choice (MS SQL, Oracle, MYSQL, PostGresQL), including customer data. My aim is simply to have incoming calls identified (using CID) and logged, then initiating a request - from the relevant client pc - to the webserver for a specific page (the customer history/details page). Now my current line of thinking is this: Install asterisk; Connect to telephone line - can I use a simple PCI modem on a standard POTS line for testing purposes?; Configure for routing incoming calls to the client machines using VOIP - this means the client machines (on a lan) need a headset with mic and can be used instead of a telephone?; Configure for CID value to cross-reference customer database, then (somehow) instruct the client machine to request the relevant customer details page. As I'm unable to implement http-push due to client machines running mainly MSIE, I was thinking, even a basic messenger pop-up displaying customer name and number might suffice; I'd also like the ability to perform http-get and post operations directly from the asterisk box as the web server is in a remote location (via VPN); I realise I could simply use a smaller module to pull out the CID but, frankly I'm interested in the PBX functionality too for future development. Is asterisk the right tool for me and am I thinking along the right lines with the above? TIA, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limiting out going calls to a maximum duration
I want to limit my sons phone useage, by setting a 30min limit on out going calls from his room is there a simple way of doing this with asterisk? id make him have a specific context, and set absolute timeout in it as 30 minutes (1800 seconds) [son-outbound] exten = _X.,1,AbsoluteTimeout,1800 exten = _X.,2,Dial,however you are dialing out... i think this should end the call after 1800 seconds... duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 CallerID
Which H.323 channel driver are you running? I don't have any H.323 endpoints ever leaving me voicemail, but I know chan_h323 does now deal with CallerID. I cannot speak for the 3rd party driver. Jeremy McNamara Rattana BIV wrote: Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some questions about a potential usage scenario for asterisk
Hi Dave, I think it will. We have a very similar requirement... My aim is simply to have incoming calls identified (using CID) and logged, then initiating a request - from the relevant client pc - to the webserver for a specific page (the customer history/details page). We're working towards exactly the same. Connect to telephone line - can I use a simple PCI modem on a standard POTS line for testing purposes?; No. An ISDN card is your quickest way to get started. Beyond that you'll need a card from Digium. Configure for routing incoming calls to the client machines using VOIP - this means the client machines (on a lan) need a headset with mic and can be used instead of a telephone?; No problem. You'll need a softphone on each PC such as X-Lite or may prefer to get a POTS card from Digium and keep standard handsets. Configure for CID value to cross-reference customer database, then (somehow) instruct the client machine to request the relevant customer details page. As I'm unable to implement http-push due to client machines running mainly MSIE, I was thinking, even a basic messenger pop-up displaying customer name and number might suffice; We're doing this with a simple web-service providing tray-app on the client PC. Asterisk can call Perl scripts so we're having one which will call the appropriate client PC and pass the caller ID. The tray app then fires up IE with the CLID in the URL. I'd also like the ability to perform http-get and post operations directly from the asterisk box as the web server is in a remote location (via VPN); Shouldn't be a problem. I realise I could simply use a smaller module to pull out the CID but, frankly I'm interested in the PBX functionality too for future development. Is asterisk the right tool for me and am I thinking along the right lines with the above? Very much. Can't recommend it highly enough. W ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Some questions about a potential usage scenario for asterisk
Hi Scott, Thanks for clarifying those few items for me. It's good to know we're not the only ones doing this kind of setup. Just one more question. You said I need at least an isdn card. So I need to have an isdn line then? forgive my ignorance, I know so little about telephony I actually feel ashamed. Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie question - devices
Title: RE: [Asterisk-Users] newbie question - devices Santiago: Just internally speaking for 20 users with very little room for growth you could purchase a T100P (T1 card) from Digium. Place the T100P it in the Asterisk server. Connect the T100P to a Zhone Z-Plex channel bank (or any other supported channel bank). The channel bank will break the T1 out in to 24 analog handset ports. Ports you could plug any analog phone in to. Next you worry about how to connect Asterisk up to the PSTN, using ISDN, PRI, or what ever is available in your area. It will necessitate the purchase of another card, something Digium can provide, but there are other options out there such as the ISDN cards supported under Linux. Actually if you have good bandwidth without any telephony cards you could choose PSTN access through any number of VoIP providers using SIP and IAX protocols. Hope this helps. Post a little more details and someone will jump in and lend you a hand, or contact me off list and we can discuss further. Good luck, - Matt -Original Message- From: santiago [mailto:[EMAIL PROTECTED]] Sent: Monday, August 04, 2003 11:06 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] newbie question - devices hi, I'm a newbie in this. I'm part of little company with 20 users, we need a pbx/central with access to and from the PSTN. i know that it is possible with asterisk, but i want to know which kind of devices i need, (interfaces and phones) thanks, -- santiago jos ruano rincn administracin servidores y servicios de internet red de datos universidad del cauca -BEGIN PGP MESSAGE- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org owGbwMvMwCQoeb96kq+XwjnGNbZJrGmZRbmJtpJvLwQn5pVkJqbnK3jlF79UCCpN zMtXCMrMS/6cxxWal1mWWlScmZKYopCSmqPgnFianMjF1WHPzMoA0gozUJDpLSfD /IB9Gre8ZHfZ+/BvkX4osko5wPfUQ4b5ST8VT3hciP3inJl578O17DedDS9fAgA= =5oc0 -END PGP MESSAGE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Some questions about a potential usagescenario for asterisk
bottom quote=on On Mon, 2003-08-04 at 11:04, Dave Wilson wrote: Hi Scott, Thanks for clarifying those few items for me. It's good to know we're not the only ones doing this kind of setup. Just one more question. You said I need at least an isdn card. So I need to have an isdn line then? forgive my ignorance, I know so little about telephony I actually feel ashamed. Thanks, Dave What we need to know to be of more assistance to you is what country, how many stations you plan to support and what kind of telco connections you have or are planning. We get into trouble once in a while because the terms we throw around, like ISDN, may mean slightly different things in different countries. Some of us resellers (blatant plug) would like to know if you are interested in more direct support :) Howard White president - VCCH, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Some questions about a potential usage scenario for asterisk
Hi Dave- I think it was Simon, not me, who answered your questions. I would add that you might consider getting one of the Digium starter development kits. Both of the non-digital kits can connect to a single analog line (you don't need ISDN to get started). One also includes a cool USB device which you can plug a phone into. This allows you to accept calls from the outside on the X100P card, and connect to them using an analogue telephone. Great way to get started! Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Wilson Sent: Monday, August 04, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Some questions about a potential usage scenario for asterisk Hi Scott, Thanks for clarifying those few items for me. It's good to know we're not the only ones doing this kind of setup. Just one more question. You said I need at least an isdn card. So I need to have an isdn line then? forgive my ignorance, I know so little about telephony I actually feel ashamed. Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Some questions about a potential usage scenario for asterisk
Thanks guys, I'm in the UK and initially I'm looking to set up a test system supporting 4 - 10 users. The test location is currently serviced by BT, with a plain analogue line and ADSL. Our actual deployment, however may be based in several european countries (UK, Ireland, Germany and Italy initially) and in several disparate client locations. We provide automotive retail management software using an ASP model. I'm guessing though, that asterisk would allow us to maintain a certain degree of standardisation in our service offering, whilst keeping our options open for supporting various types of setups to suit our clients (dealership networks). There is the other hurdle of clients with existing PBX systems in place. I've no idea how we'll cover this scenario as I'm sure most clients will be reluctant to replace their existing systems, unless of course asterisk can be plugged into some of these systems?!? TIA, Dave P.S. Howard, I'll keep the consultancy services in mind :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Syntax for hiding caller ID but still passing ANI?
I have a question regarding the flags for hiding caller ID presentation: My customer has a requirement that they are able to specify if outbound calls (on a T100P) will have the caller ID displayed or not. This could be easily solved, of course, by not setting a caller ID when creating the outbound call. However, the PRI to which this T100P is connected _must_ see a valid caller ID, and the caller ID is used for billing purposes. I know that there is the ability to hide caller ID within the Zaptel libraries, using the presentation flags. If set correctly, the expected behavior would be that the ANI would be sent to the switch, but with a flag that would tell the remote switch to suppress the caller ID from being transmitted to the end user. How does one activate that presentation switch from within a dialplan? Searching the archives gives me some hints, but no answers. Searching the code, I see in channels/chan_zap.c around line 1399 that the PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN and PRES_NUMBER_NOT_AVAILABLE are referenced, but I'm not clear on where l is set, or even if that is a trigger. Can someone give me a hand on syntax on this? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Some questions about a potential usage scenario for asterisk
There is the other hurdle of clients with existing PBX systems in place. I've no idea how we'll cover this scenario as I'm sure most clients will be reluctant to replace their existing systems, unless of course asterisk can be plugged into some of these systems?!? Yes, it can. If the PBX in question has provisions for attaching an analogue line or extension, then asterisk can be connected to the PBX. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Musiconhold interrupted sound
Hi again, Am I really the only one who's having this problem? Music on hold playing like this is very annoying and practically unusable. One more detail. The sound interruption happens only when 2 endpoints are actually connected and one of them put the other on hold. If I set a special extension with SetMusicOnHold application it seems to play just fine. Please help! Thank you. Michael On Friday 01 August 2003 04:58 pm, Michael Ulitskiy wrote: Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes then getting worse again. I've noticed that the following messages look directly connected to sound interruptions: on asterisk console: NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible in debug log: Aug 1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe The music stops playing simultenuously with these messages appearing. Also I've tried to install ztdummy and zaprtc drivers (no zaptel hardware installed). None of them helped. Asterisk version is CVS-07/30/03-18:53:16 If anybody has any idea on how to fix it or what can be done to further troubleshoot it, I'd appreciate hearing from you. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie question - devices
RE: [Asterisk-Users] newbie question - devicesHi, So let me understand this better. Asterisk can use SIP gateways which offer PSTN access. For example www.iconnecthere.com, can be used? Is this correct? And if it is, than any incoming calls through that service, could be redirected by astrisk to its users? Yes, asterisk can talk to several SIP gateways for both incoming and outgoing calls. There are also a few providers in the US that provide native IAX call origination and termination. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Musiconhold interrupted sound
Use asterisk-oh323. It works great! Michael. Michael Ulitskiy wrote: Hi again, Am I really the only one who's having this problem? Music on hold playing like this is very annoying and practically unusable. One more detail. The sound interruption happens only when 2 endpoints are actually connected and one of them put the other on hold. If I set a special extension with SetMusicOnHold application it seems to play just fine. Please help! Thank you. Michael On Friday 01 August 2003 04:58 pm, Michael Ulitskiy wrote: Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes then getting worse again. I've noticed that the following messages look directly connected to sound interruptions: on asterisk console: NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible in debug log: Aug 1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe The music stops playing simultenuously with these messages appearing. Also I've tried to install ztdummy and zaprtc drivers (no zaptel hardware installed). None of them helped. Asterisk version is CVS-07/30/03-18:53:16 If anybody has any idea on how to fix it or what can be done to further troubleshoot it, I'd appreciate hearing from you. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF modes and external IVR systems over ISDN
From: Martin Pycko [EMAIL PROTECTED] type on your asterisk CLI pri debug span spanno and send the trace of a broken call Martin, this is the trace made with debug on pri span of a call *not* recognized as answered by *. It's quite long, i'm sorry but i thought to capture all the debug infos. Enabled debugging on span 1 -- Executing SetCallerID(SIP/finetti-sip-cf27, 199770783) in new stack -- Executing Dial(SIP/finetti-sip-cf27, Zap/g1/803121||r) in new stack -- Making new call for cr 32793 Protocol Discriminator: Q.931 (8) len=32 Call Ref: len= 2 (reference 25/0x19) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Display (len= 1) [ Display (len= 1) [ 1 Display (len= 1) [ 1 ] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '803121' ] Sending Complete (len= 0) -- Called g1/803121 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32793/0x8019) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (Channel Identification) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32793/0x8019) (Terminator) Message type: PROGRESS (3) Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 30 (Progress Indicator) The debug stops here, cause for over than 20 seconds it remained with the last line displayed and no activities. on the CLI it remains the voice CLI called g1/803121 Usually, when the number is really ringing and answered, after the Called line, appears the ZapX-X is ringing line, that doesn't appear here. Thanks again, Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie question - devices
Title: RE: [Asterisk-Users] newbie question - devices Santiago: Ok then you can use asterisk as the gateway between the PSTN and an internal VoIP network. I assume you do not want to purchase any analog phones or VoIP phones, just PCs with a good sound card, speakers and a microphone? You did not clarify if your internal users were running Linux or Windows. For Linux GnoPhone is an excellent PC based phone client and it speaks IAX, a very light weight VoIP protocol just for Asterisk. By the way if you want go with VoIP phones, the Snom phone is the only hardware VoIP phone I know of that speaks IAX and lots of people are out there using it now. For Windows I have used MSN messenger 4.7, SIP and GSM codec and get so so performance from that combination. There are lots of other ones out there that will speak to Asterisk using H323 and SIP. I just do not know what they are cause I have no big need for them. You still have to connect yourself to the PSTN through your phone provider of choice in your location. In Columbia I am not sure who that would be and what type of service you can get. T1 vs E1 for example. Perhaps someone on the list can help you out in that respect. Any Columbian Asterisk users out there? - Matt -Original Message- From: santiago [mailto:[EMAIL PROTECTED]] Sent: Monday, August 04, 2003 12:51 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] newbie question - devices thanks for the answer, we need to use the data network for the transport of the voice, with the pcs as telephone devices, with h323 (possibly), and can interact with the PSTN (there is not VoIP providers here) thanks again, On Mon, 2003-08-04 at 11:51, Senad Jordanovic wrote: Hi, So let me understand this better. Asterisk can use SIP gateways which offer PSTN access. For example www.iconnecthere.com, can be used? Is this correct? And if it is, than any incoming calls through that service, could be redirected by astrisk to its users? Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of McAughan, Matt Sent: 04 August 2003 17:19 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] newbie question - devices Santiago: Just internally speaking for 20 users with very little room for growth you could purchase a T100P (T1 card) from Digium. Place the T100P it in the Asterisk server. Connect the T100P to a Zhone Z-Plex channel bank (or any other supported channel bank). The channel bank will break the T1 out in to 24 analog handset ports. Ports you could plug any analog phone in to. Next you worry about how to connect Asterisk up to the PSTN, using ISDN, PRI, or what ever is available in your area. It will necessitate the purchase of another card, something Digium can provide, but there are other options out there such as the ISDN cards supported under Linux. Actually if you have good bandwidth without any telephony cards you could choose PSTN access through any number of VoIP providers using SIP and IAX protocols. Hope this helps. Post a little more details and someone will jump in and lend you a hand, or contact me off list and we can discuss further. Good luck, - Matt -Original Message- From: santiago [mailto:[EMAIL PROTECTED]] Sent: Monday, August 04, 2003 11:06 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] newbie question - devices hi, I'm a newbie in this. I'm part of little company with 20 users, we need a pbx/central with access to and from the PSTN. i know that it is possible with asterisk, but i want to know which kind of devices i need, (interfaces and phones) thanks, -- santiago jos ruano rincn administracin servidores y servicios de internet red de datos universidad del cauca -BEGIN PGP MESSAGE- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org owGbwMvMwCQoeb96kq+XwjnGNbZJrGmZRbmJtpJvLwQn5pVkJqbnK3jlF79UCCpN zMtXCMrMS/6cxxWal1mWWlScmZKYopCSmqPgnFianMjF1WHPzMoA0gozUJDpLSfD /IB9Gre8ZHfZ+/BvkX4osko5wPfUQ4b5ST8VT3hciP3inJl578O17DedDS9fAgA= =5oc0 -END PGP MESSAGE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- santiago jos ruano rincn administracin servidores y servicios de internet red de datos universidad del cauca -BEGIN PGP MESSAGE- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org owGbwMvMwCQoeb96kq+XwjnGNbZJrGmZRbmJtpJvLwQn5pVkJqbnK3jlF79UCCpN zMtXCMrMS/6cxxWal1mWWlScmZKYopCSmqPgnFianMjF1WHPzMoA0gozUJDpLSfD /IB9Gre8ZHfZ+/BvkX4osko5wPfUQ4b5ST8VT3hciP3inJl578O17DedDS9fAgA= =5oc0 -END PGP MESSAGE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Musiconhold interrupted sound
Michael, With all due respect to both of you, it's not related to h.323 driver. The result is the same whether h.323 channel participates in the call or it's pure sip-to-sip call. Michael On Monday 04 August 2003 01:59 pm, Michael Manousos wrote: Use asterisk-oh323. It works great! Michael. Michael Ulitskiy wrote: Hi again, Am I really the only one who's having this problem? Music on hold playing like this is very annoying and practically unusable. One more detail. The sound interruption happens only when 2 endpoints are actually connected and one of them put the other on hold. If I set a special extension with SetMusicOnHold application it seems to play just fine. Please help! Thank you. Michael On Friday 01 August 2003 04:58 pm, Michael Ulitskiy wrote: Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes then getting worse again. I've noticed that the following messages look directly connected to sound interruptions: on asterisk console: NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible in debug log: Aug 1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe The music stops playing simultenuously with these messages appearing. Also I've tried to install ztdummy and zaprtc drivers (no zaptel hardware installed). None of them helped. Asterisk version is CVS-07/30/03-18:53:16 If anybody has any idea on how to fix it or what can be done to further troubleshoot it, I'd appreciate hearing from you. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie question - devices
James, Thanks for that! Do you have a list of those US AIX providers? Also, is there anyone in UK providing termination/origination on AIX? Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Sharp Sent: 04 August 2003 18:59 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] newbie question - devices RE: [Asterisk-Users] newbie question - devicesHi, So let me understand this better. Asterisk can use SIP gateways which offer PSTN access. For example www.iconnecthere.com, can be used? Is this correct? And if it is, than any incoming calls through that service, could be redirected by astrisk to its users? Yes, asterisk can talk to several SIP gateways for both incoming and outgoing calls. There are also a few providers in the US that provide native IAX call origination and termination. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM codec
The gsmfr codec works great with asterisk, all the g7* codec don't. Luciano -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Tamas Jalsovszky Enviado el: Sabado 2 de Agosto del 2003 09:38 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] GSM codec Hello, are the gsm codecs in Cisco as53xx or 36xx (gsm, gsmfr) copatible with Asterisk's gsm codec? Thanks in advance, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bridged trunks stuck off hook.
Hi, I'm getting a situation where 2 of my trunks (loopstart pots) are occationally bridged together (3624 Newbridge channel bank - asterisk signalling=fxs_ks for trunks) and are staying off hook until I do a 'soft hangup' on one of them. When I listen on a butt set each of the lines are silent (or at least by the time I find them). I am guessing one of the end users isn't properly waiting for dial tone and inadvertently doing a transfer. The line connected to Zap/1 seems most apt to be involved - I'm wondering if the telco is not indicating hangup from the remote end. Any suggestions? Maybe a timeout on bridged trunks? John snips... hc-ast*CLI show channels Channel (ContextExtensionPri ) State Appl. Data ... Zap/1-1 (hc_trunks s1 ) Up Bridged Call Zap/2-1 Zap/2-1 (hc_fxs 6000 1 ) Up Dial Zap/g1/7828503 6 active channel(s) This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Musiconhold interrupted sound
Quoting Michael Ulitskiy [EMAIL PROTECTED]: Hi again, Am I really the only one who's having this problem? Music on hold playing like this is very annoying and practically unusable. Hi Michael, No - you're not alone :) I've had similar problems with SIP since I started experimenting with * nearly a year ago. I still get it on my current cvs system every now and then although it's not nearly as bad as it used to be. Sometimes the moh will start off ok but then seem to lose sync and break up after a minute or so, other times (like now ;) ) it seems perfect apart from the odd little glitch. I just put it down to the fact that I have no real zap devices (chan_capi sip only) and so rely on ztdummy for timing. I was hoping that when I got around to putting an digium fxs card in, the problem would disappear completely! I may be wrong, but I seem to remember that Michael Manousos' oh323 channel driver was _not_ affected by this problem (this was before Jeremy McNamara's h323 channel was usable), maybe because it doesn't use the same * libraries as other voip channels, however it's been a while since I played with h323 so I don't know if that's still the case. Jamie Neil Versado I.T. Services Ltd. One more detail. The sound interruption happens only when 2 endpoints are actually connected and one of them put the other on hold. If I set a special extension with SetMusicOnHold application it seems to play just fine. Please help! Thank you. Michael On Friday 01 August 2003 04:58 pm, Michael Ulitskiy wrote: Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes then getting worse again. I've noticed that the following messages look directly connected to sound interruptions: on asterisk console: NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible in debug log: Aug 1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe The music stops playing simultenuously with these messages appearing. Also I've tried to install ztdummy and zaprtc drivers (no zaptel hardware installed). None of them helped. Asterisk version is CVS-07/30/03-18:53:16 If anybody has any idea on how to fix it or what can be done to further troubleshoot it, I'd appreciate hearing from you. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bugs.digium.com
Is anyone else having trouble accessing it with something besides IE on a Windows box? Opera on Mac/FreeBSD/Linux just hangs at the login page, IE on Mac and Netscape on Solaris Linux explode when loading login_page.php. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] bugs.digium.com
Hehe, that's slightly ironic... =P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sharp Sent: Monday, August 04, 2003 3:16 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] bugs.digium.com Is anyone else having trouble accessing it with something besides IE on a Windows box? Opera on Mac/FreeBSD/Linux just hangs at the login page, IE on Mac and Netscape on Solaris Linux explode when loading login_page.php. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bugs.digium.com
On Monday 04 August 2003 02:16 pm, James Sharp wrote: Is anyone else having trouble accessing it with something besides IE on a Windows box? Opera on Mac/FreeBSD/Linux just hangs at the login page, IE on Mac and Netscape on Solaris Linux explode when loading login_page.php. No, it works just fine on both Mozilla on Linux and Konqueror on Linux. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Musiconhold interrupted sound
On Monday 04 August 2003 02:56 pm, Jamie Neil wrote: Quoting Michael Ulitskiy [EMAIL PROTECTED]: Hi again, Am I really the only one who's having this problem? Music on hold playing like this is very annoying and practically unusable. Hi Michael, No - you're not alone :) Glad to hear. Then there's a hope it's gonna be fixed :) I've had similar problems with SIP since I started experimenting with * nearly a year ago. I still get it on my current cvs system every now and then although it's not nearly as bad as it used to be. Sometimes the moh will start off ok but then seem to lose sync and break up after a minute or so, other times (like now ;) ) it seems perfect apart from the odd little glitch. I just put it down to the fact that I have no real zap devices (chan_capi sip only) and so rely on ztdummy for timing. I was hoping that when I got around to putting an digium fxs card in, the problem would disappear completely! The strange thing is that the fact whether ztdummy is loaded or not does not really affect sound quality in my case. I guess asterisk might not using/seeing it. Does it requires some special configuration? I may be wrong, but I seem to remember that Michael Manousos' oh323 channel driver was _not_ affected by this problem (this was before Jeremy McNamara's h323 channel was usable), maybe because it doesn't use the same * libraries as other voip channels, however it's been a while since I played with h323 so I don't know if that's still the case. This sounds really strange to me as I think the problem is central to asterisk, not to particular driver and behavior is exactly the same over sip and chan_h323. Ok. I'll try chan_oh323 and see if it makes any difference. Jamie Neil Versado I.T. Services Ltd. Thanks Jamie. Michael One more detail. The sound interruption happens only when 2 endpoints are actually connected and one of them put the other on hold. If I set a special extension with SetMusicOnHold application it seems to play just fine. Please help! Thank you. Michael On Friday 01 August 2003 04:58 pm, Michael Ulitskiy wrote: Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes then getting worse again. I've noticed that the following messages look directly connected to sound interruptions: on asterisk console: NOTICE[245776]: File rtp.c, Line 239 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible in debug log: Aug 1 16:43:58 DEBUG[32771]: File res_musiconhold.c, Line 292 (monmp3thread): Only wrote -1 of 640 bytes to pipe The music stops playing simultenuously with these messages appearing. Also I've tried to install ztdummy and zaprtc drivers (no zaptel hardware installed). None of them helped. Asterisk version is CVS-07/30/03-18:53:16 If anybody has any idea on how to fix it or what can be done to further troubleshoot it, I'd appreciate hearing from you. Thanks. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callwaiting in sip can't be disabled
On Mon, 2003-08-04 at 14:31, Brian West wrote: What type of phones? Grandstream BudgeTones. Is it a function of the phones? Is there any way to limit them in sip.conf to one channel each? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Repost MS Messenger 4.7 docs?
I had M$ Mess working a bit ago but now I cant seem to make it work. can someone post whatI need in SIP.conf for a comfig to get it working again? Thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM codec
On Mon, 4 Aug 2003, Luciano Ramos wrote: The gsmfr codec works great with asterisk, all the g7* codec don't. huh?? g711 works fine, too (both A and µ). But maybe you don't count them as codecs... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zhone Zplex 10 units
On Mon, 2003-08-04 at 13:13, John Schmerold wrote: I frequently see Zhone Zplex 10 units on Ebay - cheap. What's the story on these? Are they flaky? search the archives. Tough to configure? tough, no, pain in the ***, yes Any other issues that come to mind? search the archive, that is why it is there. I don't see them listed on Zhone's website (except in support), so I suspect they've discontinued the product, but if it's a good product I could use it to learn Asterisk. Thats funny since they don't really even act like they want to support them. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Test
Test ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM codec
I've tested g711ulaw and alaw in cisco AS5300 and they don't work with asterisk, gsmfr in AS5350 works great. Luciano -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Siggi Langauf Enviado el: Lunes 4 de Agosto del 2003 19:02 Para: Asterisk user list Asunto: RE: [Asterisk-Users] GSM codec On Mon, 4 Aug 2003, Luciano Ramos wrote: The gsmfr codec works great with asterisk, all the g7* codec don't. huh?? g711 works fine, too (both A and µ). But maybe you don't count them as codecs... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault
When i make a call from x100p to quicknet
-- Executing Dial(Zap/1-1, Phone/phone0|5) in new stack
Segmentation fault
extension file
[quicknet]
exten =3D _1XX,1,Dial(Zap/0/${EXTEN})
phone file
[interfaces]
mode=3Ddialtone
format=3Dslinear
echocancel=3Dmedium
context=3Dquicknet
device =3D /dev/phone0
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[Asterisk-Users] Fix for Redhat 9 zombie AGI processes
Hi all- Thanks to Mark Spencer for finding this patch: If you are experiencing leftover zombie processes from your AGI scripts that have terminated, this is apparently due to a RedHat 9 threading issue introduced in a recent update To get rid of this, try entering the following line before you start asterisk: export LD_ASSUME_KERNEL=2.4.1 It works for me - I'm still checking to see if there are any other side effects - none yet.. -Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP clients not sending audio
Found the problem, it looks like if you set 'allow=all' in sip.conf then audio doesn't work. Commented it out and it all works beautifully. Adam Adam Donnison wrote: Hi, I've got two SIP clients, one is X-Lite on NT, the other is KPhone on Linux and when I try either the echo test or voicemail demos, they fail to send any audio. They are both set up as type of friend in sip.conf so that they can send and receive calls. Using an IAX client like Gnophone, I have no problems. The troubling thing is that I'm almost certain that this was working prior to the latest CVS version. I've also played around with the config files so it is just as likely going to be my fault. I've run tcpdump and can see the call establishment, and the voice data coming from * to the SIP client, but then there is nothing from the SIP client back to *, which seems a little strange. Can anyone suggest anything, or at least a direction in which to look? Adam -- Adam Donnison email: [EMAIL PROTECTED] Saki Computer Services Pty. Ltd. 93 Kallista-Emerald Roadphone: +61 3 9752 1512 THE PATCH VIC 3792AUSTRALIAfax: +61 3 9752 1098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Cisco 7960, SIP, NAT, Voicemal
Try this: exten = 4001,1,Dial(SIP/gadams,10,r) I don't know how the syntax you've specified will behave; maybe it will work, but it's not any format I've used. Try the syntax above for your Dial line and see if it results in different actions. Plus, I'm slightly confused as to your dialplan. When a call comes in from your 7960, it is passed to context ATL. Now, when you call your 7960, where is that call going? Are you passing your inbound and outbound calls both to context ATL? If not, then you're missing an important part of the debug (and perhaps having a conceptual problem with how call flow works.) JT -Original Message- From: Adams, Gavin Sent: Monday, August 04, 2003 6:10 PM To: '[EMAIL PROTECTED]' Subject: Cisco 7960, SIP, NAT, Voicemal Hey all, I've got a couple 79xx phones working peer-to-peer and am now trying to work on the voice mail. In extensions.conf: [ATL] exten = 4001,1,Dial(SIP/gadams)|10 exten = 4001,2,Voicemail,u4001 exten = 4001,102,Voicemail,b4001 and the corresponding sip.conf: [gadams] type=friend username=gadams secret=** context=ATL host=dynamic canreinvite=no nat=yes mailbox=4001 When this phone is dialed, it doesn't roll over to VM after 10 seconds but continues to ring. If the calling party hangs up, the phone continues to ring. However, as a test I changed the |10 to a |10t. At that point dialing 4001 did indeed roll over to voicemail, but it happened immediately. Also, I'm getting the following message during the dial: WARNING[1133735216]: File chan_sip.c, Line 417 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Which is tied to the call in question. Any clues? --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Syntax for hiding caller ID but stillpassing ANI?
OK, I can understand that, and I suppose that your answer perhaps indicates that there is currently no method within Asterisk that is easily accessible that permits such tagging of outbound PRI calls. Putting such a patch or feature together is beyond my ability. Ultimately, I'd like to set this flag from within the dialplan somehow such that the call is flagged with that option upon creation. I suppose I'll put this on my little pile of things that would be nice to have, but I can't afford. :) I did a little digging, and found that there is even an RFC Draft for SIP extensions for privacy that translates to PRI signalling, which would be nice to have, though certainly all I'm shooting for is a flag to set in the dialplan (through a SetVar command maybe?) Anyway, if anyone gets ambitious about handling it, here's the well-written Cisco documentation on how they implement a version of it: SIP Extensions for Caller Identity and Privacy http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t13/ftsipext.htm#1027188 It seems that there are two options that block the presentation: 0x23 and 0x21 JT l is set a couple of lines above. Basically l carries the nubmer so if there is no callerid in 'l' then we send this other flag 'callerid not available'. You need to choose one of these flags: /* Presentation */ #define PRES_ALLOWED_USER_NUMBER_NOT_SCREENED 0x00 #define PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN 0x01 #define PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN 0x02 #define PRES_ALLOWED_NETWORK_NUMBER 0x03 #define PRES_PROHIB_USER_NUMBER_NOT_SCREENED0x20 #define PRES_PROHIB_USER_NUMBER_PASSED_SCREEN 0x21 #define PRES_PROHIB_USER_NUMBER_FAILED_SCREEN 0x22 #define PRES_PROHIB_NETWORK_NUMBER 0x23 #define PRES_NUMBER_NOT_AVAILABLE 0x43 I think it might be PROHIBPASSED_SCREEN but not sure. Check q931 specs. Martin On Mon, 4 Aug 2003, John Todd wrote: I have a question regarding the flags for hiding caller ID presentation: My customer has a requirement that they are able to specify if outbound calls (on a T100P) will have the caller ID displayed or not. This could be easily solved, of course, by not setting a caller ID when creating the outbound call. However, the PRI to which this T100P is connected _must_ see a valid caller ID, and the caller ID is used for billing purposes. I know that there is the ability to hide caller ID within the Zaptel libraries, using the presentation flags. If set correctly, the expected behavior would be that the ANI would be sent to the switch, but with a flag that would tell the remote switch to suppress the caller ID from being transmitted to the end user. How does one activate that presentation switch from within a dialplan? Searching the archives gives me some hints, but no answers. Searching the code, I see in channels/chan_zap.c around line 1399 that the PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN and PRES_NUMBER_NOT_AVAILABLE are referenced, but I'm not clear on where l is set, or even if that is a trigger. Can someone give me a hand on syntax on this? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fix for Redhat 9 zombie AGI processes
Having similar problems with Debian Woody. Is this a known issue or merely poorly written AGI scripts from my side? Fredrik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: den 5 augusti 2003 00:39 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fix for Redhat 9 zombie AGI processes Hi all- Thanks to Mark Spencer for finding this patch: If you are experiencing leftover zombie processes from your AGI scripts that have terminated, this is apparently due to a RedHat 9 threading issue introduced in a recent update To get rid of this, try entering the following line before you start asterisk: export LD_ASSUME_KERNEL=2.4.1 It works for me - I'm still checking to see if there are any other side effects - none yet.. -Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie question - devices
On Mon, Aug 04, 2003 at 12:50:34PM -0500, santiago wrote: MESSAGE with PGP'd part Please do not post PGP encrypted messages to any mailing list. A PGP signature depends on your key, not the recipients. Those would be ok. -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callwaiting in sip can't be disabled
On Mon, 2003-08-04 at 22:41, Steve Meyers wrote: What type of phones? Grandstream BudgeTones. Is it a function of the phones? Is there any way to limit them in sip.conf to one channel each? Looking at the source of channels/chan_sip.c, the threewaycalling parameter seems not to be honoured. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
[Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, questions about call transfers
hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3 seconds delayed to the speech .. there is no echo on incoming voice, just an echo of my own voice as I speak. 2nd question: using a grandstream phone asterisk, if I hear another phone ringing, how can answer it from the phone infront of me? eg. if extension 6003 is ringing, and i have phone number 6004, how can I answer it ? 3rd question: can someone give me some starter hints to configure call parking ? I haven't managed to find a direct way to transfer a call from phone to phone except using blind transfer and I want the person initiating the transfer to speak to the receiving person before actually passing the call. can anybody help please ? cheers Dave A Caruana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP calls cause segmentation fault
does anyone of the programmers know if this has been fixed in a more recent CVS version? should I redownload and recompile? cheers Dave - Original Message - From: Adam Donnison [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 1:18 AM Subject: Re: [Asterisk-Users] SIP calls cause Segmentation Fault I actually found this same thing, and traced it down to app_dial.c line 190. It doesn't explicitly check for a valid chan before trying to use it and it segfaults when it does a strlen on a chan entity. I simply put a check in that winner was non-zero before comparing it to o-chan: if (winner winner == o-chan) Adam Dave Alan Caruana wrote: I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan is a modification of the sample one that came with Asterisk, and I haven't touched that many other conf files actually. Any way I can get this debugged? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Adam Donnison email: [EMAIL PROTECTED] Saki Computer Services Pty. Ltd. 93 Kallista-Emerald Roadphone: +61 3 9752 1512 THE PATCH VIC 3792AUSTRALIAfax: +61 3 9752 1098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transitioning from existing PBX
Hello - Warning: newbie questions at 12 o'clock! I am looking at transitioning my house from a Panasonic KXT1232 to an asterisk-based PBX (*PBX). We currently have 5 CO lines and 16 extensions plus ADSL with a static IP. I have looked at the various docs, but find I still have some questions. 1. Interface cards a. Should I buy a T100P + channel bank? (BTW: Why isn't this what is in the developer's kit, instead of the T400P?) b. If I am going to simply replace the Panasonic and supply 2-wire FXS (is that correct terminology?) to the existing jacks, can I use the T100P with (Z-Plex 10? model A or B?) and have 8 FXO and 16 FXS? I think so, but the developer's kit only references FXS c. Or must I buy more equipment, e.g. 4 X100P's + 4 TDM400Ps?. d. Are there other combinations worth considering? e. Others to be avoided? 2. To prevent disruption to the rest of the household, I would like start out with the *PBX behind the KXT -- i.e. the *PBX's CO lines are actually extensions on the KXT. In this situation, is the *PBX (or Z-Plex) seeing FXS or FXO on its CO lines? I would guess FXS, but I'm confused. 3. If the *PBX is receiving an in-bound IP call, can it present it to the KXT as another CO line? Is this trivial or postdoctoral stuff? 4. Any suggestions about the best way to convert 2-wire analog phones to IP-based? Cisco? Grandstream? Xnet? Others? 5. We have 8 Panasonic 4-wire phones with Lots O' Buttons (tm). What should I consider for replacements? ADSI? IP-phones? Soft IP phones? Any opinions about features vs. price vs. voice quality vs. reliability? (I'm sure there are! :-) ) Thanks in advance! Peter (When I spell-checked this message, Netscape suggested epiphanies as a replacement for IP-phones. :-) ) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP calls cause segmentation fault
Yes, latest CVS fixes this. Adam Dave Alan Caruana wrote: does anyone of the programmers know if this has been fixed in a more recent CVS version? should I redownload and recompile? cheers Dave -- Adam Donnison email: [EMAIL PROTECTED] Saki Computer Services Pty. Ltd. 93 Kallista-Emerald Roadphone: +61 3 9752 1512 THE PATCH VIC 3792AUSTRALIAfax: +61 3 9752 1098 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel banks, etc.
If I understand this correctly.. Your channel bank will convert analog FXO FXS to a T1 then you slap a Wildcard T100P in your linux box then your all set. It shouldn't matter if the channel bank is compatible with linux or not because you are going to terminate with a T1 cross over to your linux box. bkw On Mon, 4 Aug 2003, Steve Meyers wrote: Where can I find a good tutorial on how channel banks work? I need to get a 6 port (or so) channel bank for FXO. I need to find some information on which ones are supported well under Linux and with Asterisk, how to configure them, what specifically to look for in a channel bank, etc. I'm pretty new to all this, so I'm not familiar with a lot of the terms and such. Thanks! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel banks, etc.
Where can I find a good tutorial on how channel banks work? I need to get a 6 port (or so) channel bank for FXO. I need to find some information on which ones are supported well under Linux and with Asterisk, how to configure them, what specifically to look for in a channel bank, etc. I'm pretty new to all this, so I'm not familiar with a lot of the terms and such. Thanks! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM codec
Does your IOS on the AS5300 support g711ulaw? bkw On Mon, 4 Aug 2003, Luciano Ramos wrote: I've tested g711ulaw and alaw in cisco AS5300 and they don't work with asterisk, gsmfr in AS5350 works great. Luciano -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Siggi Langauf Enviado el: Lunes 4 de Agosto del 2003 19:02 Para: Asterisk user list Asunto: RE: [Asterisk-Users] GSM codec On Mon, 4 Aug 2003, Luciano Ramos wrote: The gsmfr codec works great with asterisk, all the g7* codec don't. huh?? g711 works fine, too (both A and µ). But maybe you don't count them as codecs... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
