Re: [Asterisk-Users] Location of Voice e-mail Code???
I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. On Sat, Mar 12, 2005 at 01:03:00PM +0300, Julius Kidubuka wrote: Hi, Where can I find the code that performs the voice e-mail function (that is, the code that reads the contents of voicemail.conf and then performs the necessary action)? I am using [EMAIL PROTECTED] 0.6. The mail is delivered by piping it to a sendmail program (by default /usr/sbin/sendmail). /usr/sbin/sendmail does not have to be sendmail. Postfix and Exim provide a sendmail-compatible interface along with a host of more minimal programs such as ssmtp and nullmailer. With sendmail and similar (Exim and Postfix) the aliases (normally /etc/aliases) file is a useful place to set up forwarding. e.g: suppose you want to keep your voicemail.conf as simple as possible: [default] #vmbox=pass,name,recipients 200=200,,[EMAIL PROTECTED] 201=201,,[EMAIL PROTECTED] 202=202,,[EMAIL PROTECTED] 202=202,,[EMAIL PROTECTED] 203=203,,[EMAIL PROTECTED] 204=204,,[EMAIL PROTECTED] to your aliases file you could then add: 200: john 201: [EMAIL PROTECTED] 202: david,|/usr/local/bin/send_sms_to_david Note that I have ommited the names, but those names are actually also used for things other than voicemail. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail SMS Alert - Possible?
I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. ___ 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
R: [Asterisk-Users] Voicemail SMS Alert - Possible?
Use externnotify (see http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) with a script to send sms. Some time ago I used a perl script called sendSms found in Internet. Bye. Marco -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Julius Kidubuka Inviato: lunedì 14 marzo 2005 09.09 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Voicemail SMS Alert - Possible? I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom 220 busy all the time
Good day all We have a snom 220 that for some reason keeps on giving this message Got SIP response 486 Busy Here back from 192.168.21.222 even though there is no active calls to it and there are 2 accounts set on the phone? Please Help and advice Thanks Altus ___ 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] Text Messaging or AIM
quote who=C F Why not, just use the email address given before from you email client. So, you can generate an SMS message on your Cell phone and send it to your, say, hotmail account? Or are you talking about using an embeded email client on the phone to create an email. Not using SMS at all? I have my asterisk box setup in voicemail.conf to send me notifications to me cell phone using this method. I'm a Sprint subscriber (I used to be Verizon, and it worked with them as well), so I have * setup to send me an email to [EMAIL PROTECTED], letting me know there is a new voicemail waiting for me, callerID of caller, and duration and in which mailbox. I can then call back my * box and listen to the messages, I like this better than the callback feature b/c I can do it on my time. This is easy. Just put the phonenum@carrier.com address in the definition of the voicemail box in voicemail.conf. Though this is not the direction I have been talking about. -- END OF LINE -MCP ___ 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] Text Messaging or AIM
quote who=Eric Wieling Robert Hajime Lanning wrote: um, backwards. E-Mail to SMS. I have not seen the other way around. Both Cingular and Verizon supports both. I have not tried this, nor have I seen any documentation mentioning it. Do you or anyone else have a pointer for the info? Especially for Cingular, as that is what I am with, currently. -- END OF LINE -MCP ___ 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] EADS6550 and asterisk - echo on PSTN call
Rich Adamson a écrit : would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side, give echo. We can do what we want in zconfig.h (STEVE2, MARK2, MMX, AGGRESSIV_SUPPRESSOR, NOECHOCAN_DISABLE) or zapata.conf (tx gain=-10.0, echocancel=32 ... 256,), test with differents POTS phone, it change nothing. We even didn't notice changes between our various changes in those files (and yes modules where unloaded between each test). Always the same echo. So know we start to doubt that this echo problem is asterisk related but perhaps more to the PBX. That's why we ask if some of you have/had similar setup with this PBX and if there is a solution. You didn't mention what country your in; if you outside the US, be sure to config the TDM-fxo card for your country (eg, line impedance). France. You mention echocancel=32, etc, did you try echotraining=800? Yes. It create a second echo :-( For my TDM-fxo in the US, using the following on each channel works fine: echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=5.0 txgain=0.0 Do you have this setup with the standard zconfig.h (MARK2)? I might have missed some of your earlier posts relative to this; just catching up on over 500 emails from this list. I've not had to configure a TDM for non-US support, but I know for an absolute fact (based on 20 years of detailed telephony engineering experience) that you have to config the TDM card for line impedance, etc, for your country. If you've not done that, start there. (Think that's an optional parameter when loading the drivers.) I update asterisk from cvs-head about every two weeks or so, and always stick with default values (including zconfig.h). So, yes I'm using the default echo cancellation, etc. There has not been very many changes associated the the zaptel source code and the TDM-fxo drivers. Certainly not necessary to use the latest cvs-head at all; anything from the last few months should work. My /etc/zaptel.conf is adapted to country: loadzone=fr defaultzone=fr Asterisk stable 1.0.5. If you're telling that I have to pass parameters to module when loading, I checked with modinfo wctdm (at office I have head version) and options I have are those: [EMAIL PROTECTED] asterisk]# /sbin/modinfo -p wctdm debug int loopcurrent int robust int _opermode int opermode string timingonly int lowpower int boostringer int fxshonormode int battdebounce int battthresh int alawoverride int Pardon my ignorance but no one of them remaind me to impedance. And for what I saw earlier in the source file, those informations could be updated with the value of the zaptel.conf file. I believe its the opermode string that needs to be set to a country. Not sure what values are acceptable, but one google result indicated: opermode=Australia as an example. The driver name for the tdm-fxo card/modules has changed to wctdm, so when you look at those google examples, keep that in mind. I pretty sure you need to do the same thing for the TDM-fxs card. Thanks to you and Richard. I add the opermode and fxshonormode, it help to reduce it strongly. I compile with MMX stuff and MARK2. Will try to add agressive cancellation. -- Daniel ___ 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] Panasonic KX-TD1232
Title: Panasonic KX-TD1232 Hello, I'm trying to connect an Asterisk server with the Panasonic KX-TD1232 Phone System. Is this possible? Which hardware do I need and which Asterisk configuration files should I adjust? Dennie __This mail has been scanned for all known viruses by AXSWeb powered by SecuTeam NV. _ This mail has been scanned for all known viruses by AXS Mail powered by SecuTeam NV. Register for AXS Mail at http://www.secuteam.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: R: [Asterisk-Users] Voicemail SMS Alert - Possible?
Another option is to send sms by mail. 1-) You subscribe to an sms provider who can allow you to do mail2sms; 2-) You send sms message under the form [EMAIL PROTECTED] ; 3-) SMS provider receives SMS from you and will send it through its gateway; Hope this helps Quoting Marco Ziglioli [EMAIL PROTECTED]: Use externnotify (see http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) with a script to send sms. Some time ago I used a perl script called sendSms found in Internet. Bye. Marco -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Julius Kidubuka Inviato: lunedì 14 marzo 2005 09.09 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Voicemail SMS Alert - Possible? I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] weird outbound problem through broadvoice (new)
Hello, Have a weird problem when using asterisk (1.0.6). There are certain numbers I cannot dial when using asterisk with my broadvoice account. No problems with inbound. With outbound calls, I can call some numbers (for example broadvoice customer support number) and unsuccessfully with some. However, when I configure my account directly on x-lite, I dont see these outbound problems. Here is a snapshot of my sip.conf register = [EMAIL PROTECTED]:PP:[EMAIL PROTECTED] [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromuser=UU fromdomain=sip.broadvoice.com secret=PP username=UU port=5060 dtmfmode=inband dtmf=inband insecure=very context=incoming authname=UU canreinvite=no qualify=no nat=no extensions.conf [outgoing] exten = _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) exten = _1NXXNXX, 2, congestion() exten = _1NXXNXX, 102, busy() A portion of sip debug during successful calls (calling broadvoice support) Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a From: 1001 sip:[EMAIL PROTECTED];tag=as65b65920 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE 6 headers, 0 lines CLI Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a From: 1001 sip:[EMAIL PROTECTED];tag=as65b65920 To: sip:[EMAIL PROTECTED];tag=SD58a8499-104694000-1110784950009 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY Supported: 100rel,timer Contact: sip:[EMAIL PROTECTED]:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp Remote-Party-ID: Auto Attendant PrimaryAttendantsip:[EMAIL PROTECTED];user=phone;bvoice=ACME-06t5tpji5ub7e;screen=yes;party=called;privacy=off;id-type=subscriber Content-Length: 0 A portion of sip debug during unsuccessful calls, where T is the target phone number Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 From: 1001 sip:[EMAIL PROTECTED];tag=as6f6dba69 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE 6 headers, 0 lines Reliably Transmitting: CANCEL sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 From: 1001 sip:[EMAIL PROTECTED];tag=as6f6dba69 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 CANCEL User-Agent: Asterisk PBX Proxy-Authorization: Digest username=UU, realm=BroadWorks, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=1110785211206, response=f68a31735aec843b9ef68b7909fcf178, opaque= Content-Length: 0 (no NAT) to 147.135.8.128:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Transmitting (no NAT): SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK01853115f3033a3c From: sip:[EMAIL PROTECTED];tag=9d9e03fd7b4508e9 To: sip:[EMAIL PROTECTED];tag=as79fd7936 Call-ID: [EMAIL PROTECTED] CSeq: 7327 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to x.x.x.x:5060 Asterisk box not behind firewall. No iptables filters either. It seems that asterisk is sending CANCEL due to call timeout after the 2nd 100 Trying during INVITE message flow. I am not sure what is causing the timeout. Anyone experienced this before? Tried using ethereal to debug the problem deeply, but I can only see the same flow as the sip debug. Hoping for your assistance. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Voicemail SMS Alert - Possible?
We do have that service here using [EMAIL PROTECTED] The way it works is that I can do mail2sms and sms2mail. What I would like to do is to have my * box send an sms to a cellphone, that is, to say [EMAIL PROTECTED] where 0485.. is my cellphone number and it.co.ug my sms provider/domain. This sms should be sent as soon as I get a voicemail message in my mail inbox. A kind of sms e-mail alert service. Hope this is much clearer. Otherwise thanks for all the contributions so far. Still waiting for more... Rgds, Julius. Another option is to send sms by mail. 1-) You subscribe to an sms provider who can allow you to do mail2sms; 2-) You send sms message under the form [EMAIL PROTECTED] ; 3-) SMS provider receives SMS from you and will send it through its gateway; Hope this helps Quoting Marco Ziglioli [EMAIL PROTECTED]: Use externnotify (see http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) with a script to send sms. Some time ago I used a perl script called sendSms found in Internet. Bye. Marco -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Julius Kidubuka Inviato: lunedì 14 marzo 2005 09.09 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Voicemail SMS Alert - Possible? I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Asterisk Billing System
On Fri, 2005-03-11 at 20:18, Kanishka Somaratne wrote: Hi Is there a billing system that i can view all the call taken by SIP clients in asterisk http://www.voip-info.org/tiki-index.php?page=Asterisk+CDR+Areski+GUI __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mitel working together with Asterisk??
I am looking at the possibility integrating Asterisk with our current Mitel 200sx. If this is possible what physical connection is made between the Mitel box and * box? Then can a user choose if a call is go out VoIP or not? I'm more familiar with the SX2000 family rather than the 200 series, but what you want to do IS possible. You could connect with a T1 between the Mitel and an Asterisk box.. or even with some FXO/FXS ports if you didn't want such a large scale implementation. Feel free to contact me off-list if you want to discuss further. Cheers Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail SMS Alert - Possible?
Have a look at this site http://www.bayhamsystems.com/asterisk.html It was easy to install and the example works fine. No word on commercial pricing yet but you can test in the meantime. Ian Julius Kidubuka wrote: I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Panasonic KX-TD1232
On Mon, 14 Mar 2005, Dennie Verstrepen wrote: I'm trying to connect an Asterisk server with the Panasonic KX-TD1232 Phone System. Is this possible? Which hardware do I need and which Asterisk configuration files should I adjust? Yes, it is possible. How it is done depends on what interfaces you have in the Panasonic at the moment and what you are willing to spend to acheive various levels of integration. One option is to connect using an E1 between Asterisk (e.g. a Digium TE410P card) and a KX-TD290 E1 card on the Panasonic. This will make Asterisk look like the PSTN to the Panasonic. The same can be acheived using the BRI lines on the baseboard in the KX-TD1232. Another option is to hook up Asterisk as analogue extensions, but this is a lot less flexible. Unfortunatly Panasonic only offers an integrated dialplan between pbx:es with their TIE cards which use few analogue connections. We use a KX-TD290 to a TE405P card and then another E1 to the PSTN. Peter ___ 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] asterisk 1.0.6
Hi after upgrade from R2 to 1.0.6 , my dtmf not working and i cant dial . can any 1.0.6 user help me why i cant do that. Bashir ___ 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] Panasonic KX-TD1232
Hi Peter, I'm looking at a similar set up using a GHX1232 but I can't find a single refence or docmentation for a GHX1232 anywhere though, and I'm a bit of a newbie to this game. Do you know if it would take a similar approach to integrate asterisk into that system? Thanks John - Peter Svensson wrote: On Mon, 14 Mar 2005, Dennie Verstrepen wrote: I'm trying to connect an Asterisk server with the Panasonic KX-TD1232 Phone System. Is this possible? Which hardware do I need and which Asterisk configuration files should I adjust? Yes, it is possible. How it is done depends on what interfaces you have in the Panasonic at the moment and what you are willing to spend to acheive various levels of integration. One option is to connect using an E1 between Asterisk (e.g. a Digium TE410P card) and a KX-TD290 E1 card on the Panasonic. This will make Asterisk look like the PSTN to the Panasonic. The same can be acheived using the BRI lines on the baseboard in the KX-TD1232. Another option is to hook up Asterisk as analogue extensions, but this is a lot less flexible. Unfortunatly Panasonic only offers an integrated dialplan between pbx:es with their TIE cards which use few analogue connections. We use a KX-TD290 to a TE405P card and then another E1 to the PSTN. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.0.5 and h323 compiling problem
Hello! Looks like h323 compiling is FAQ, but I didn't found an answer... The same problem with 0.6.5 and 0.7.1: gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/var/local/files/asterisk-1.0.5/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1454: error: structure has no member named `cid' chan_oh323.c:1454: error: structure has no member named `cid' chan_oh323.c:1456: error: structure has no member named `cid' chan_oh323.c:1468: error: structure has no member named `cid' chan_oh323.c:1470: error: structure has no member named `cid' chan_oh323.c:1470: error: structure has no member named `cid' chan_oh323.c:1472: error: structure has no member named `cid' chan_oh323.c:1484: error: structure has no member named `cid' And there is really no cid in ast_channel ... How can I compile h323 with Asterisk 1.0.5? Thank you! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
On Sun, 13 Mar 2005 21:49:52 +0200, Dimitris Kounalakis [EMAIL PROTECTED] wrote: Hello *Martijn, Thank you for your response. *That was my opinion too, it looses the context due to a bug, and can anyone confirm it also? But I have no output from the command Show channels, and it happens so quickly that it is impossible to issue the command before falling to the default context. In the logs, I can see that the channel exists like CAPI[contr1/2810211694]/0 but this is druring call only. Any other way to debug it more (or to solve it)? My /etc/asterisk/capi.conf is: - [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] controller=1 msn=2810111694 incomingmsn=* devices=2 softdtmf=1 callgroup=1 context=isdn On my system I have the devices=2 as the last line this works for me [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=330417 incomingmsn=* controller=1 softdtmf=1 accountcode= context=from-pstn echocancel=yes echotail=64 devices=2 ___ 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] Log Error
On Sun, Mar 13, 2005 at 07:22:58PM -0600, Anton Krall wrote: So far nobody has answered this post... Anybody has seen this error before? Could you use a more verbose logging? IIRC, the technology is the channel type, e.g: sip, zap, iax. Somewhere something is getting either an empty channel name or some garbage as the channel name. [trimmed bottom-posting] -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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 TE405P and Slackware 10.0
Hi to all, I have a problem with this wildcard and one E1 line. The server is a Asus P4P800S with the 865PE chipset, and 512MB RAM. The kernel version is 2.4.26. I have donwload and build the latest CVS version of zaptel, libpri and asterisk following the ufficial instructions of digium. I have set up my server in this way: zaptel.conf loadzone=it defaultzone=it span=1,1,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 zapata.conf switchtype = EuroISDN signalling = pri_cpe pridialplan = unknown context = incoming group = 2 channel = 1-15,17-31 [EMAIL PROTECTED]:/usr/src# rmmod wct4xxp zaptel [EMAIL PROTECTED]:/usr/src# modprobe zaptel [EMAIL PROTECTED]:/usr/src# modprobe wct4xxp [EMAIL PROTECTED]:/usr/src# ztcfg -vv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. [EMAIL PROTECTED]:/usr/src# lsmod Module Size Used by Not tainted wct4xxp 51680 0 (unused) zaptel 175904 0 [wct4xxp] [EMAIL PROTECTED]:/usr/src# cat /proc/pci Bus 2, device 9, function 0: Communication controller: PCI device 10ee:0314 (Xilinx Corporation) (rev 1). IRQ 3. Master Capable. Latency=64. Non-prefetchable 32 bit memory at 0xf7eef800 [0xf7eef87f]. [EMAIL PROTECTED]:/usr/src# cat /proc/interrupts CPU0 0: 115864 XT-PIC timer 1: 141 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC t4xxp 5: 2583 XT-PIC eth0 8: 1 XT-PIC rtc 10: 0 XT-PIC Intel ICH5 14: 2639 XT-PIC ide0 NMI: 0 ERR: 0 [EMAIL PROTECTED]:/usr/src# dmesg Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 3 for device 02:09.0 Found TE410P at base address f7eef800, remapped to e0978800 TE410P version c01a009b, burst ON FALC version: 0005, Board ID: 00 Reg 0: 0x1eec6800 Reg 1: 0x1eec6000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 11 (Italy) TE410P: Span 1 configured for CCS/HDB3/CRC4 SPAN 1: Primary Sync Source TE410P: Span 1 configured for CCS/HDB3/CRC4 SPAN 1: Primary Sync Source Registered tone zone 11 (Italy) The jumpers are on E1, and the crossed cable should be ok. Without loading the module the LED glows in red colour, but the moment we load module, it goes off. (No red or green). We ran zttool and tried to run a loop test, but zttool simply hung with the message 'Looping UP Span 1...'. We had to terminate zttool with 'kill'. What's wrong ? Thanks, pixer ___ 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] E1/T1 back to back ??
Hi there Just a quick question, I will be building some servers in a lab utilizing Digium E1 cards. I would like if possible to avoid the expense of installing an e1/ISDN30 in my lab. I have two questions really, first does anybody know of an effective simulation tool I can use to replicate a real world PRI but without the telco line being installed. And secondly, can I have a scenario with 2 asterisk servers with digium e1 cards 'back to back' one configured as the network side and the other configured as the client side (can I just use a single cat5 straight through cable between them ?? and cant the Digium e1 cards operate ok in both modes?) Any advice would be greatly appreciated ___ 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 TE405P and Slackware 10.0
Hi to all, I have a problem with this wildcard and one E1 line. The server is a Asus P4P800S with the 865PE chipset, and 512MB RAM. The kernel version is 2.4.26. I have donwload and build the latest CVS version of zaptel, libpri and asterisk following the ufficial instructions of digium. I have set up my server in this way: zaptel.conf loadzone=it defaultzone=it span=1,1,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 zapata.conf switchtype = EuroISDN signalling = pri_cpe pridialplan = unknown context = incoming group = 2 channel = 1-15,17-31 [EMAIL PROTECTED]:/usr/src# rmmod wct4xxp zaptel [EMAIL PROTECTED]:/usr/src# modprobe zaptel [EMAIL PROTECTED]:/usr/src# modprobe wct4xxp [EMAIL PROTECTED]:/usr/src# ztcfg -vv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. [EMAIL PROTECTED]:/usr/src# lsmod Module Size Used byNot tainted wct4xxp51680 0 (unused) zaptel175904 0 [wct4xxp] [EMAIL PROTECTED]:/usr/src# cat /proc/pci Bus 2, device 9, function 0: Communication controller: PCI device 10ee:0314 (Xilinx Corporation) (rev 1). IRQ 3. Master Capable. Latency=64. Non-prefetchable 32 bit memory at 0xf7eef800 [0xf7eef87f]. [EMAIL PROTECTED]:/usr/src# cat /proc/interrupts CPU0 0: 115864 XT-PIC timer 1:141 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC t4xxp 5: 2583 XT-PIC eth0 8: 1 XT-PIC rtc 10: 0 XT-PIC Intel ICH5 14: 2639 XT-PIC ide0 NMI: 0 ERR: 0 [EMAIL PROTECTED]:/usr/src# dmesg Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 3 for device 02:09.0 Found TE410P at base address f7eef800, remapped to e0978800 TE410P version c01a009b, burst ON FALC version: 0005, Board ID: 00 Reg 0: 0x1eec6800 Reg 1: 0x1eec6000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 11 (Italy) TE410P: Span 1 configured for CCS/HDB3/CRC4 SPAN 1: Primary Sync Source TE410P: Span 1 configured for CCS/HDB3/CRC4 SPAN 1: Primary Sync Source Registered tone zone 11 (Italy) The jumpers are on E1, and the crossed cable should be ok. Without loading the module the LED glows in red colour, but the moment we load module, it goes off. (No red or green). We ran zttool and tried to run a loop test, but zttool simply hung with the message 'Looping UP Span 1...'. We had to terminate zttool with 'kill'. What's wrong ? Thanks, pixer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk outbound to SIP provider problems
Hi I am having alot of difficulty connecting to SIP providers (I am trying 3) and can't seem to find anything similar in the wiki or on the lists.I can receive inbound calls fine however on placing an outbound call the calling phone never gets 'connected' but 2 way audio is passed for about 20secs before some sort of timeout. Anything suggestions as to what I could try appreciated. Many thanks Walt. -- The call goes like this: caller: dial caller: SIP code 100 destination: ring caller: 1-2 second delay caller: SIP code 183 caller: ring destination: pickup caller: 2 way audio ok destination: 2 way audio ok caller: Sip code 183 (Never 200) caller: some sort of call timout, audio stops destination: chooses to hang up caller: chooses to hang up sip debug peer of a provider: http://www.walt.9k.com/sip/1_SIP_Provider.html sip debug peer of phone placing the call http://www.walt.9k.com/sip/1_cisco_phone.html _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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] Re: Sipura 841 issues
Master Abi [EMAIL PROTECTED] wrote: Not having a backlit display is bad design. Actually it is a feature issue, not a design issue. :) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.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] N/A
Hello!! Please help me with next problem... While traying to read voicemail system plays all service messages and then hang upthe line... console display next: Mar 14 14:16:02 WARNING[135271424]: file.c:1004 ast_waitstream_full: Wait failed (No such file or directory) Asterisk runing on FreeBSD 5.0 -- KTURE WebMail ___ 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] asterisk outbound to SIP provider problems
Hi Good Morning, I am having alot of difficulty connecting to SIP providers (I am trying 3) and can't seem to find anything similar in the wiki or on the lists.I can receive inbound calls fine however on placing an outbound call the calling phone never gets 'connected' but 2 way audio is passed for about 20secs before some sort of timeout. I'm having the same problem over here, but with both, inbound/outbound calls, I use a SER server to auth my users, and when I need to use a VoIP line that is not at my server, I use Asterisk to auth the line outside my server at my Foreign Voip server then when I get the line I can dial, but none of them, incoming/outgoing, calls are working fine. How did you configure your incoming call ? The call goes like this: caller: dial caller: SIP code 100 destination: ring caller: 1-2 second delay caller: SIP code 183 caller: ring destination: pickup caller: 2 way audio ok destination: 2 way audio ok caller: Sip code 183 (Never 200) caller: some sort of call timout, audio stops destination: chooses to hang up caller: chooses to hang up sip debug peer of a provider: http://www.walt.9k.com/sip/1_SIP_Provider.html sip debug peer of phone placing the call http://www.walt.9k.com/sip/1_cisco_phone.html _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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 -- Felipe Martins Mundivox Communications Tecnologia e Projetos [EMAIL PROTECTED] Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.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] asterisk codec negotiation problem
hello list, i searched for nearly a week for a solution to this problem, as there is: analog fax machine -» grandstream ata -» asterisk -» sip trunk from provider -» provider gateway to pstn -» analog/isdn fax machine on pstn everything worked out fine until my provider decided to implement t38 into the gateway. now when i send/receive a fax message the gateway tries to connect with t38 and waits for a media capability unknown to receive if the device on the sip trunk is incapable of talking t38, in order to fall back to the old g711 behaviour. but asterisk does not answer with media capability unknown (sorry, did forget the sip message number), he answers with his own codec capabilities, and this is ignored by the gateway. so he tries to send with t38 anyway, and that of course fails. any thougts on that one? kind regards, michael ___ 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] Re: Panasonic KX-TD1232
Hi, Dennie, Yes, it is possible. Specially that model has DTMF Inband signaling, in other words you can send dtmf tones to asterisk when using it as a voicemail so that it knows what extension did not answer the call and can thus be directed to the right voicemail. You need to play with the programming on both the pbx and the asterisk box. I dont know what kind of integration you need, but assuming you want to add voicemail to the pbx, my recomendation is to have all incoming lines from your telcom connected to the pbx. Have the lines terminate on 4 extentions which in turn are conected to a 4 port fxo digium card. Configure the extentions on the pbx to be forwarded to that group of 4 lines if they are busy or an-answered. Hope it helps. -- Sergio Veltri www.pointhorizon.com mail: [EMAIL PROTECTED] Tel: +5411-5217-1295 Cell: +54-911-5604-4149 I'm trying to connect an Asterisk server with the Panasonic KX-TD1232 Phone System. Is this possible? Which hardware do I need and which Asterisk configuration files should I adjust? Dennie ___ 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] Panasonic KX-TD1232
On Mon, 14 Mar 2005, John Brennan wrote: I'm looking at a similar set up using a GHX1232 but I can't find a single refence or docmentation for a GHX1232 anywhere though, and I'm a bit of a newbie to this game. Do you know if it would take a similar approach to integrate asterisk into that system? I have never heared of a 'GHX1232' so I am afraid I cannot help you there. Peter ___ 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] E1/T1 back to back ??
On Mon, 14 Mar 2005, Brett, Gary wrote: Just a quick question, I will be building some servers in a lab utilizing Digium E1 cards. I would like if possible to avoid the expense of installing an e1/ISDN30 in my lab. I have two questions really, first does anybody know of an effective simulation tool I can use to replicate a real world PRI but without the telco line being installed. And secondly, can I have a scenario with 2 asterisk servers with digium e1 cards 'back to back' one configured as the network side and the other configured as the client side (can I just use a single cat5 straight through cable between them ?? and cant the Digium e1 cards operate ok in both modes?) You can put two Asterisk-boxes back-to-back. Configure one as pri_cpe and one as pri_net. You need an E1 cross over cable which is different from an ethernet cross over cable. Search the net for which pins to connect. Other alternatives exist. They may be closer to what you would see from a real pstn connection, but they are also a lot more expensive. Peter ___ 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] Panasonic KX-TD1232
Thats cool Peter, thanks, Has anyone on the list ever heard of a Goldstar GHX-1232? It seems to be a bit of a dinosaur. I'm hoping it might be a rebranded device though and someone might be able to point me in the right direction for documentation? Thanks in advance. - connect IT tel:(+353)(0) 1 4099703 fax:(+353)(0) 1 4099793 mob:(+353)(0) 86 8296611 e-mail: [EMAIL PROTECTED] www:www.connectIT.ie Peter Svensson wrote: On Mon, 14 Mar 2005, John Brennan wrote: I'm looking at a similar set up using a GHX1232 but I can't find a single refence or docmentation for a GHX1232 anywhere though, and I'm a bit of a newbie to this game. Do you know if it would take a similar approach to integrate asterisk into that system? I have never heared of a 'GHX1232' so I am afraid I cannot help you there. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with TE405P and Slackware 10.0
On March 14, 2005 06:50 am, pixer wrote: 3: 0 XT-PIC t4xxp Without loading the module the LED glows in red colour, but the moment we load module, it goes off. (No red or green). We ran zttool and tried to run a loop test, but zttool simply hung with the message 'Looping UP Span 1...'. We had to terminate zttool with 'kill'. What's wrong ? This is a hardware or BIOS issue -- your card is unable to generate interrupts. Try shuffling the card around to a different PCI slot and/or adjusting your BIOS interrupt settings. Also you might want to try the pci=noacpi or even noapic kernel options. I run the same card in a similar box with Slackware 10.0 (and formerly 9.1) without any issue whatsoever. This is a specific hardware issue. -A. ___ 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] E1/T1 back to back ??
On March 14, 2005 06:43 am, Brett, Gary wrote: Just a quick question, I will be building some servers in a lab utilizing Digium E1 cards. I would like if possible to avoid the expense of installing an e1/ISDN30 in my lab. I have two questions really, first does anybody know of an effective simulation tool I can use to replicate a real world PRI but without the telco line being installed. And secondly, can I have a scenario with 2 asterisk servers with digium e1 cards 'back to back' one configured as the network side and the other configured as the client side (can I just use a single cat5 straight through cable between them ?? and cant the Digium e1 cards operate ok in both modes?) A standard Cat5 ethernet cable won't work, but a T1/E1 crossover cable made from Cat5 should work just fine. I do this all the time with T1/PRI, I don't see why it wouldn't work with E1/PRI. One side is set up as pri_cpe, and hte other as pri_net. -A. ___ 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] PRI Call Reference Length not Supported
check your entensions.conf file /etc/asterisk/extensions.conf
. ${ETEN:${TRUNKMSD}})
we had same problem this was the fix
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Sunday, March 13, 2005 1:23 PM
Subject: [Asterisk-Users] PRI Call Reference Length not Supported
Using CVS-HEAD libpri, CVS-HEAD zaptel, CVS-STABLE asterisk.
Everything compiled fine. No problems loading chan_zap.so.
Incomming calls to PRI work fine. Outbound is a different story:
-- Executing Dial(SIP/64.72.107.4-4122fb40, ZAP/R1d/18005551212|60)
in new stack
-- Called R1d/18005551212
-- Channel 0/23, span 1 got hangup
Mar 13 13:19:29 WARNING[28835]: chan_zap.c:7149 zt_pri_error: PRI: Call
Reference Length not supported: 0
-- Zap/23-1 is circuit-busy
-- Hungup 'Zap/23-1'
== Everyone is busy/congested at this time
I've never experienced this before. Anyone have any ideas? I'm going to
revert back to STABLE versions of libpri and zaptel and see if I get the
same error.
-Matthew
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RE: [Asterisk-Users] Cisco and Asterisk
Hi!
First of all , (apart from solving your problem) you really should get
rid of the whole [demo] context from extensions.conf, and place your
stuff in your own context (i.e. [local]) (just for convenience and
security). Getting back to the problem - as I see it you want to dial
out through Cisco gw by dialing 1XXX
To do so you must send the whole number to the gateway, so the gateway
could do something (anything) with it.
Your extensions.conf should be more like:
exten = _1XXX,1,Dial(SIP/[EMAIL PROTECTED])
Dialing 1602 on your system-phone will result with sending the number
1602 to the gateway, which will then (according to your current
dial-peer configuration) strip the leading 1 and send 602 back to
Asterisk to dial your laptop.
Hope it helps
Brgs
Tomek
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ben Miller
Sent: Friday, March 11, 2005 12:40 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco and Asterisk
Hey all,
I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can
get
a bit of help here.
First I'll explain my setup, and then my problem.
Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2
FXO
ports. I have an analog phone line plugged into the first port
(voice-port 1/0/0). I've got it setup so that calls coming into that
analog line are transferred to my Asterisk server via SIP.
In the second port on my FXO card, I have a phone cable plugged into a
phone-system phone (the kind you have in the office plugged into your
phone system, the extra port on it acts as an FXS so a normal phone
can
be plugged into it and can dial out by hitting 9,9 and then a number).
Incoming calls come into my * box fine, and I can hit digits on the
phone and have different thing happen. For example, I setup XLite on
my
work laptop and I've got an extension setup to dial my laptop. What
I'm
trying to do, though, is setup an extension that will connect back to
my
router and let me make an outgoing call on the second voice port.
Every
time I try to do this, I get SIP errors in the * CLI:
Got SIP response 400 Bad Request - 'Malformed/Missing URL' back from
206.222.200.46.
206.222.200.46 is the IP of my router. I'm pretty sure that I'm just
missing some config in my router, but I've been googling the past few
days and can't get anything that's helping. Thus, I turn to you to
help
me out, if possible.
I work for an ISP and what we eventually want to do is setup VoIP for
our broadband customers so they can do unlimited dialing to various
cities where we have routers, and we'll just through some voice ports
into those routers and get some lines hooked up.
Here is my relevant config:
sip.conf:
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
dtmfmode=inband
nat=never
promiscredir = yes ; If yes, allows 302 or REDIR to non-local SIP
address
[voice-gw]; This is what I've setup for my Cisco
; has the voice ports
context=demo
type=friend
host=206.222.200.46 ; IP address of Cisco gateway
dtmfmode=inband
disallow=all
allow=ulaw
nat=no
qualify=yes
[ben] ; my work laptop
context=demo
type=friend
username=ben
host=dynamic
disallow=all
allow=ulaw
extensions.conf:
[general]
static=yes
writeprotect=no
; You can include other config files, use the #include command
(without
the ';')
; Note that this is different from the include command that includes
contexts within
; other contexts. The #include command works in all asterisk
configuration files.
;#include filename.conf
; The Globals category contains global variables that can be
referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for
Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface
for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than General and Globals represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
;
Re: [Asterisk-Users] E1/T1 back to back ??
The cross-over cable is what I do between by Asterisk and my Lucent PBX's. Works great!! Peter Svensson wrote: On Mon, 14 Mar 2005, Brett, Gary wrote: Just a quick question, I will be building some servers in a lab utilizing Digium E1 cards. I would like if possible to avoid the expense of installing an e1/ISDN30 in my lab. I have two questions really, first does anybody know of an effective simulation tool I can use to replicate a real world PRI but without the telco line being installed. And secondly, can I have a scenario with 2 asterisk servers with digium e1 cards 'back to back' one configured as the network side and the other configured as the client side (can I just use a single cat5 straight through cable between them ?? and cant the Digium e1 cards operate ok in both modes?) You can put two Asterisk-boxes back-to-back. Configure one as pri_cpe and one as pri_net. You need an E1 cross over cable which is different from an ethernet cross over cable. Search the net for which pins to connect. Other alternatives exist. They may be closer to what you would see from a real pstn connection, but they are also a lot more expensive. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIPJet and g.711
Just in my dial plan. I am not using any real Lease cost routing package, as a matter of fact I am developing one but it's not ready yet. W - Original Message - From: Robert Augustyn [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, March 13, 2005 6:21 PM Subject: RE: [Asterisk-Users] VoIPJet and g.711 Thanks, Are you doing it by setting the lowest cost? Is there anything in Asterisk which does it? Thanks, robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wojciech Tryc Sent: Sunday, March 13, 2005 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoIPJet and g.711 Robert, Nufone, but it all depends on the destination. For some is gafachi, for some is VoicePulse etc.. W - Original Message - From: Robert Augustyn [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com; 'Justin Richards' [EMAIL PROTECTED] Sent: Sunday, March 13, 2005 12:09 PM Subject: RE: [Asterisk-Users] VoIPJet and g.711 Wojtek, What are you using for your primary route? robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wojciech Tryc Sent: Sunday, March 13, 2005 9:31 AM To: Justin Richards; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoIPJet and g.711 I can see errors on the console, g.729 and ilbc works no problem. I endup moving VoIPjet to the secondary route. Wojtek - Original Message - From: Justin Richards [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 12, 2005 11:00 PM Subject: Re: [Asterisk-Users] VoIPJet and g.711 I am having problem with voipjet and g.711 (ulaw) as well. I tried ilbc with no luck. basically my outbound call connects, i can hear them talk, but they can't hear me. i am not getting errors in console with either ulaw or ilbc, just no audio to the called party. it worked great yesterday, and I haven't changed anything.. my connection to voicepulse (same settings ad voipjet) works great. On Fri, 11 Mar 2005 12:33:14 -0500, Wojciech Tryc [EMAIL PROTECTED] wrote: I am experiencing problems connecting to VoIPjet with g.711. It works with g.729 and ilbc. It used to work... Anyone? Regards, Wojtek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Cisco 7960 SIP 7.4
For those that are interested, I was just out on the Cisco site and noticed that they had released firmware 7.4 as of March 11th for the 7940/7960 phones. Doug ___ 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] colinux fresh install, zaptel does not compile, size_t error
Hi, I followed the instructions on http://www.asterisk.org/index.php?menu=download. I picked the latest version using CVS. Things went fine until I cd zaptel ; make clean ; make install. I then get an error when compiling zaptel.c /usr/src/linux/include/linux/kernel.h:75: error: parse error before size_t This happens very early on and I suspect that it is actually an issue with the kernel include files on my machine. Nota: I am installing on a colinux debian. uname -a Linux colinux2 2.4.26-co-0.6.1 #1 Sat May 29 15:30:37 IDT 2004 i686 GNU/Linux I think I have all the required packages, but I maybe wrong. If anybody else has had the same issue, thanks for help. PS: I could not find a way to search the mailing list archive... Yours, JeanHuguesRobert PS: dpkg -l output: colinux2:/usr/src/zaptel# dpkg -l Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed |/ Err?=(none)/Hold/Reinst-required/X=both-problems (Status,Err: uppercase=bad) ||/ NameVersion Description +++-===-===-== ii adduser 3.56 Add and remove users and groups ii apt 0.5.25 Advanced front-end for dpkg ii apt-utils 0.5.25 APT utility programs ii at 3.1.8-11 Delayed job execution and batch processing ii base-files 3.0.15 Debian base system miscellaneous files ii base-passwd 3.5.7 Debian base system master password and group files ii bash2.05b-15 The GNU Bourne Again SHell ii bind9 9.2.3+9.2.4-rc2-1 Internet Domain Name Server ii binutils2.15-5 The GNU assembler, linker and binary utilities ii bison 1.875d-1 A parser generator that is compatible with YACC ii bsdmainutils6.0.14 collection of more utilities from FreeBSD ii bsdutils2.12-7 Basic utilities from 4.4BSD-Lite ii bzip2 1.0.2-5 high-quality block-sorting file compressor - utilities ii coreutils 5.0.91-2 The GNU core utilities ii cpio2.5-1.1 GNU cpio -- a program to manage archives of files. ii cpp 3.3.5-1 The GNU C preprocessor (cpp) ii cpp-3.3 3.3.5-3 The GNU C preprocessor ii cron3.0pl1-83 management of regular background processing ii cvs 1.12.9-6 Concurrent Versions System ii debconf 1.4.25 Debian configuration management system ii debconf-i18n1.4.25 full internationalization support for debconf ii debconf-utils 1.4.41 debconf utilities ii debhelper 4.2.27 helper programs for debian/rules ii debianutils 2.8.2 Miscellaneous utilities specific to Debian ii defoma 0.11.8-0.1 Debian Font Manager -- automatic font configuration framework ii devscripts 2.8.5 Scripts to make the life of a Debian Package maintainer easier ii diff2.8.1-6 File comparison utilities ii dpkg1.10.22 Package maintenance system for Debian ii dpkg-dev1.10.25 Package building tools for Debian ii dselect 1.10.22 a user tool to manage Debian packages ii e2fslibs1.35-6 The EXT2 filesystem libraries ii e2fsprogs 1.35-6 The EXT2 file system utilities and libraries ii ed 0.2-20 The classic unix line editor ii equivs 2.0.6-0.1 Circumvent Debian package dependencies ii exim3.36-11 An MTA (Mail Transport Agent) ii fakeroot1.2.1 Gives a fake root environment ii fdutils 5.4-20030718-3 Linux floppy utilities ii file4.07-2 Determines file type using magic numbers ii fileutils 5.0.91-2 The GNU file management utilities (transitional package) ii findutils 4.1.20-3 utilities for finding files--find, xargs, and locate ii flex2.5.31-31 A fast lexical analyzer generator. ii fontconfig 2.2.3-4 generic font configuration library ii ftp 0.17-12 The FTP client. ii gcc 3.3.5-1 The
RE: [Asterisk-Users] Looking for a free SIP/IAX softphone with IMandpresence support
Yes, I've seen it already, but it's not really as user-friendly as sjphone. In firefly, you cannot even paste the phone number in. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Montag, 14. März 2005 02:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Looking for a free SIP/IAX softphone with IMandpresence support Firefly? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roman Zhovtulya Sent: Domingo, 13 de Marzo de 2005 04:49 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Looking for a free SIP/IAX softphone with IM andpresence support Hello, Could anyone recommend something similar in functionality and user-friendliness to SJPhone, but that would additionaly have IM and presence support? Thanks a lot, Roman Zhovtulya ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] colinux fresh install, zaptel does not compile, size_t error
[EMAIL PROTECTED] wrote: I followed the instructions on http://www.asterisk.org/index.php?menu=download. I picked the latest version using CVS. Things went fine until I cd zaptel ; make clean ; make install. I then get an error when compiling zaptel.c /usr/src/linux/include/linux/kernel.h:75: error: parse error before size_t This happens very early on and I suspect that it is actually an issue with the kernel include files on my machine. Nota: I am installing on a colinux debian. uname -a Linux colinux2 2.4.26-co-0.6.1 #1 Sat May 29 15:30:37 IDT 2004 i686 GNU/Linux On http://www.ramdyne.nl/ you can find an article on how I got rid of the same problems you were having (on a Debian sarge install). Unfortunately the server is down for the next couple of hours... Here's a link to the google cache copy: http://66.102.9.104/search?q=cache:pR1IMCaiRcQJ:www.ramdyne.nl/index.php%3Fcat%3D11+%2Basterisk+%2Bramdyne+%2Bdebianhl=nlstart=1 -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Telecom echo cancel disable
Title: Message Too hard to say. My problem is with a Channel bank, if it made it any better then it's very little. -Original Message-From: Stuart Hirst [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 09, 2005 2:51 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Telecom echo cancel disable I also have users that suffer from random echo on a British Telecom provided PRI. Can you confirm that this has improved your user experience ? Stuart -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dennis WebbSent: 09 March 2005 16:18To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Telecom echo cancel disableYeah. Edit zconfig.h and there's an option to ignore 2100hz. I didn't know what caused the 2100 until you said something.On Wed, 2005-03-09 at 09:47, Matt Schulte wrote: Disabled echo canceller because of tone (tx) on channel 10 I understand that the PSTN companies use their own echo canceller's, send a tone across 2100hz, the problem we're having is people are complaining of echo on random calls. I'm assuming this may be the cause. Is their anyway to 'ignore' the disabling of EC? Or would be just be a manual code change.. Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text Messaging or AIM
Robert Hajime Lanning wrote: quote who=Eric Wieling Robert Hajime Lanning wrote: um, backwards. E-Mail to SMS. I have not seen the other way around. Both Cingular and Verizon supports both. I have not tried this, nor have I seen any documentation mentioning it. Do you or anyone else have a pointer for the info? Especially for Cingular, as that is what I am with, currently. Send a text message. Instead of putting a naked phone number in the To: field, put in an e-mail address. At least with Cingular, you need to have text messaging enabled on your cell phone account. --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Looking for a free SIP/IAX softphone with IMandpresence support
Roman Zhovtulya wrote: Yes, I've seen it already, but it's not really as user-friendly as sjphone. In firefly, you cannot even paste the phone number in. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Montag, 14. März 2005 02:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Looking for a free SIP/IAX softphone with IMandpresence support Firefly? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roman Zhovtulya Sent: Domingo, 13 de Marzo de 2005 04:49 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Looking for a free SIP/IAX softphone with IM andpresence support Hello, Could anyone recommend something similar in functionality and user-friendliness to SJPhone, but that would additionaly have IM and presence support? Thanks a lot, Roman Zhovtulya ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You could try compiling gaimphone by Linspire. It integrates an IP soft phone with gaim. It is found at http://http://www.phonegaim.com. Hope this helps regards Kavit ___ 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] asterisk 1.0.6
I had this issue. I configured my sip phones to use rfc2293(?) instead of inband. Note:the rfc number is incorrect but I don't feel like looking up the correct one right now. Just look in sip.conf example and it will tell you the right number. On Mon, 2005-03-14 at 04:51, Bashir Ullah - www.Lamsre.Com wrote: Hi after upgrade from R2 to 1.0.6 , my dtmf not working and i cant dial . can any 1.0.6 user help me why i cant do that. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: upgrade to CVS 3/13/05, voicemail problems
WARNING[9013]: pbx.c:1554 pbx_extension_helper: No application 'VoiceMailMain2' for extension (local, 225, 1) I see now that VoiceMailMain2 has been depreciated /VoiceMail is now replaced by VoiceMail2 in the CVS, so voicemail2 will be obsolete soon. The old voicemail is not included in the current CVS. /OJ dec 2003/ sorry to bother Niles ___ 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] snom 220 busy all the time
On Monday 14 March 2005 08:50, Altus Snyman wrote: Good day all We have a snom 220 that for some reason keeps on giving this message Got SIP response 486 Busy Here back from 192.168.21.222 even though there is no active calls to it and there are 2 accounts set on the phone? Either someone has turned DND on in the phone, or the phones have not registered properly. B ___ 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] How to Flash() a modem line
Hello! I'd like to Flash() a modem line (BRI) with Asterisk. It is a passive ISDN-card connected to a hardware PBX. I use ISDN4Linux. I recognised that unfortunately the Flash() application flashes Zap devices only. Now I am wondering how I could flash Modem/ttyI0. The source code chan_modem.c doesn't contain anything about flashing a modem line. So I tried to simply put the AT-command sequence ! in my dialstring, but it didn't work. Are there any solutions? Greetings! Raoul ___ 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] Asterisk security problem: authorized SIP users can fake any callerid!
On Mon, March 14, 2005 17:06, Andres said: You might want to try the steps provided above yourself Peter. Because even if we have a context that leads to never never land at the top of sip.conf, I am still able to make free calls. A sip debug clearly Welcome to the wonderful world of stateless UDP connections... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ 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] Location of Voice e-mail Code???
I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? This is probably cheating: I have a free email account with my cell provider and they give 50 free SMS to alert when an email is received. When I'm travelling, I set this address up and I get an SMS every time a vmail is left for me. Nothing was done inside asterisk, it just sends an email to an address which handles the forwarding. ___ 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] DS3 with Asterisk
Title: Message I have done some research on the discussions that have occured on this list about DS3s with Asterisk. It seems to be dead and I have not found any active work on the project. I know that a full DS3 may have some technical limitations with why they may not work with Asterisk but I am interested in utilizing a "partial" DS3. Is there anyone utilizing DS3s out there with asterisk at all and if so how are you implementing it? (Splitting? Custom Drivers? Etc..) Has anybody attempted/failed/succeeded to make a DS3 to asterisk work? Michael ___ 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] Panasonic KX-TD1232
I have works Panasonic TD1232 with asterisk! In my office have TD1232 with one expanssion! So i have 64 extenssions, i bought a TDM04B card and connect 4 extensions of TD1232 at card. I Configure the number 8 to give me a tone of whatever 4 lines(g1) and go to internet at other asterisk with other card TDM04B with Panasonic 616, and all works!! My dialplan its very simple. You must configure the Panasonic with one group circular and choose one digit(in my case 8) for flexing number and quickdial! --all this is in panasonic. The tdm04B is configured in zapata, zaptel, sip and extenssions. I you have problem sendme an email! On Mon, 2005-03-14 at 10:28 +0100, Dennie Verstrepen wrote: Hello, I'm trying to connect an Asterisk server with the Panasonic KX-TD1232 Phone System. Is this possible? Which hardware do I need and which Asterisk configuration files should I adjust? Dennie __ This mail has been scanned for all known viruses by AXSWeb powered by SecuTeam NV. _ This mail has been scanned for all known viruses by AXS Mail powered by SecuTeam NV. Register for AXS Mail at http://www.secuteam.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 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] busy signal not in cdr
hi list. i have the following problem. if i dial an ip endpoint from my ip phone and the endpoint is busy, in my cdr i see (answered). I think there must be busy. why is that? any hints? thx, thomas ___ 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] dial script, send variable problem??
hallo,
i trying to dial with a python script via the manager interface, it works
ok but i would like to send a soud file name as a variable to the dialplan,
so that i can call a number and send it a different soundfile i choose in
my pyton script.
the problem is, that the * dials correct and sends a sound but only if its
hardcodet, the variable my script sends will not bee seen in the dialplan?
how is it possibe to send a variable via a callscript to an extention?
thanks,
alexander
-
.. python code ...
sound=feature-not-avail-line
s.send('Action: Originate\r\n')
s.send('Channel: IAX2/[EMAIL PROTECTED]/501\r\n')
s.send('Context: outboundmsg\r\n')
s.send('Extention: s\r\n')
s.send('Priority: 1\r\n\r\n')
s.send('RetryTime: 300\r\n')
s.send('WaitTime: 45\r\n')
#s.send(('Variable: snd=%s\r\n') % (sound))
s.send(('SetVar: snd=%s\r\n') % (sound))
// extentions.conf
; callgen test
[outboundmsg]
exten = s,1,DigitTimeout,5
exten = s,2,ResponseTimeout,10
exten = s,3,Answer
exten =
s,4,Wait(1)
exten = s,5,NoOp(${snd})
exten = s,6,Playback(${snd})
;exten = s,6,Playback(feature-not-avail-line)
exten = s,7,Hangup
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Re: [Asterisk-Users] Voicemail SMS Alert - Possible?
look at this thread: http://lists.digium.com/pipermail/asterisk-users/2005-March/094509.html On Mon, 14 Mar 2005 11:09:11 +0300 (EAT), Julius Kidubuka [EMAIL PROTECTED] wrote: I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for a free SIP/IAX softphonewith IMandpresence support
That looks really much closer to wht I'm looking for, but it doesn't seem to support SIP (at least Windows version). When I go to add an account, it gives only ICQ, Yahoo, MSN and a couple of others. What I would like to have is to connect it to my Asterisk via SIP. Is there any other way/any other client? Thanks a lot, Roman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kavit Munshi Sent: Montag, 14. März 2005 15:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Looking for a free SIP/IAX softphonewith IMandpresence support Roman Zhovtulya wrote: Yes, I've seen it already, but it's not really as user-friendly as sjphone. In firefly, you cannot even paste the phone number in. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Montag, 14. März 2005 02:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Looking for a free SIP/IAX softphone with IMandpresence support Firefly? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roman Zhovtulya Sent: Domingo, 13 de Marzo de 2005 04:49 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Looking for a free SIP/IAX softphone with IM andpresence support Hello, Could anyone recommend something similar in functionality and user-friendliness to SJPhone, but that would additionaly have IM and presence support? Thanks a lot, Roman Zhovtulya ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You could try compiling gaimphone by Linspire. It integrates an IP soft phone with gaim. It is found at http://http://www.phonegaim.com. Hope this helps regards Kavit ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC - Are there some add ons available?
I am trying to get more familiar with ASTCC, but I miss some tools, but I believe somebody has already thought about it: 1. I would like to send a standard letter to the users, as soon their balance drops below a certain value. E.g., Dear user, you have only 1.- left, consider to fill up... To make a php program which is started with cron every midnight should do the initial trick, however, I don't want to spam them daily, only the first time it should be send out the message. To make it simple, every Sunday we delete this record file and start over again, ... that means the first time the user get it any day, but than only every Sunday (or what day we choose) 2. I am looking for a program, that ads incoming records as well into the database, another agi after incoming call hang up??? 3. So far I could not figure out how to use BRANDS, .. since there is no connection between trunk / routes / caller and brands. Maybe there could be a parameter handled over to the agi, but I have not seen it yet. Does anybody know how to use brands? 4. The list of the card usage is in my opinion not optimized. The user, who ask for a card info, (or the admin) does not really need in each line the card number (or own caller-id), or could it be differ once? I think more important would be the date and time of the call. 5. I have not found an edit button to edit a card, only to refill, drop or reset in use. bye Ronald ___ 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] Location of Voice e-mail Code???
Take a look at this one: http://lists.digium.com/pipermail/asterisk-users/2005-March/094509.html You can also enable call back in voicemail.conf On Mon, 14 Mar 2005 11:07:49 +0300 (EAT), Julius Kidubuka [EMAIL PROTECTED] wrote: I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. Thanks, Julius. On Sat, Mar 12, 2005 at 01:03:00PM +0300, Julius Kidubuka wrote: Hi, Where can I find the code that performs the voice e-mail function (that is, the code that reads the contents of voicemail.conf and then performs the necessary action)? I am using [EMAIL PROTECTED] 0.6. The mail is delivered by piping it to a sendmail program (by default /usr/sbin/sendmail). /usr/sbin/sendmail does not have to be sendmail. Postfix and Exim provide a sendmail-compatible interface along with a host of more minimal programs such as ssmtp and nullmailer. With sendmail and similar (Exim and Postfix) the aliases (normally /etc/aliases) file is a useful place to set up forwarding. e.g: suppose you want to keep your voicemail.conf as simple as possible: [default] #vmbox=pass,name,recipients 200=200,,[EMAIL PROTECTED] 201=201,,[EMAIL PROTECTED] 202=202,,[EMAIL PROTECTED] 202=202,,[EMAIL PROTECTED] 203=203,,[EMAIL PROTECTED] 204=204,,[EMAIL PROTECTED] to your aliases file you could then add: 200: john 201: [EMAIL PROTECTED] 202: david,|/usr/local/bin/send_sms_to_david Note that I have ommited the names, but those names are actually also used for things other than voicemail. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Has anybody experience with SetGroup / CheckGroup commands?
I am checking on the SetGroup / CheckGroup commands, but I have some troubles to undestand the examples. SetGroup(moh) can be moh anything as I like? Usually moh stands for music on hold CheckGroup(1) checks if somebody in in group moh. Does it mean I can only have one SetGroup(xxx) ?? When I look at example 2 than I see two SetGroup commands and one CheckGroup command. I don't understand it!!! Can anybody explain it for me, please? I want to understand it, so that I can figure out how to setup what I need: dial with astcc a number via gateway-1 if gateway-1 is used already 1 time, than use gateway-2, and so forth, ... bye Ronald ___ 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] Panasonic KX-TD1232
I have connected the KX-1232 to asterisk with the T1 card. Is it dissappointing though as I have not gotten any Caller id information running over the T1. But it does function. Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail SMS Alert - Possible?
-Original Message- From: Julius Kidubuka [mailto:[EMAIL PROTECTED] I need to be able to send an sms alert to one's mobile/cell phone. For instance, when I receive a voicemail message in my inbox, I also want to be able to get a message on my cell phone alerting me of this e-mail. How possible is this? And if it is, what do I need to do to get the service up and running? Ideas are most welcome. If your cell phone service offers an email-to-sms gateway, putting the email address in the pager_email field of voicemail.conf works pretty well. That may be more of a U.S. thing, though. ___ 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] OT: Recommendation for Dynamic DNS on Meshbox?
I'm going to do a deployment of LocustWorld MeshBoxes in some of our remote locations. Build 90 comes with Asterisk 1.0, and our plan is to use the MeshBoxes as a WAP for non-Asterisk uses but also to add a 2nd NIC to deploy Snom's in the remote location. This works fine (was suprisingly easy to do), and I have the Meshbox running Asterisk and IAX'ing to our primary no problem. They will be on a cable ISP for the broadband that only gives dynamic IP's. I would like to use a Dynamic DNS service so we can address the boxes as foo.somewhere.com for remote managment. Does anyone have a recommendation of a free / non free etc for a Dynamic DNS service? Note the Meshbox is a brutally stripped 2.4.x kernel so deploying software to it is a problem, dependency hell and I also don't want to mess too much with the base config. Ideally, a stand alone executable for the client update or a really really well crafted shell script. I can get wget and cron running on it so I am considering writing a web service that sits on our primary end and cron can call wget and POST the output of IFCONFIG; the web service can parse out the dynamic IP from the IFCONFIG output. Another possibility is to grab the IP from the wiana.org registry; unfortunately they don't have a web service for it so it would involve a script that parses out the IP from the HTML, but that would break if wiana decides to change the HTML or if wiana discontinues operations. ___ 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] How to flash a modem line
* Mateo Meier [EMAIL PROTECTED] [2005-02-28 09:13]:
I tryed that with capi.. but no luke. It will hang up the line anyway :-(
exten = s,1,Playback(transfer)
exten = s,2,Flash(capi/72044**:041720,18)
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
Any idears why ?
No wonder! The Flash() application can flash zap channels only
and you are trying to flash a modem channel. There's no obvious
facility to flash a modem channel as I can find nothing about it
in chan_modem.c. :-\
I want to do the same thing and I am using ISDN4Linux. Perhaps we
can try submit AT-commands to the modem line. A timed H0 and H1
command would do the thing. I'm afraid we'd have to change the
asterisk source code though. It would even work with CAPI as you
can use ISDN4Linux on top of CAPI, too.
Please tell me if you find any solutions :-)
Greets!
Raoul
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Re: [Asterisk-Users] How to Flash() a modem line
Raoul Bönisch wrote: Hello! I'd like to Flash() a modem line (BRI) with Asterisk. It is a passive ISDN-card connected to a hardware PBX. I use ISDN4Linux. I recognised that unfortunately the Flash() application flashes Zap devices only. Now I am wondering how I could flash Modem/ttyI0. The source code chan_modem.c doesn't contain anything about flashing a modem line. So I tried to simply put the AT-command sequence ! in my dialstring, but it didn't work. Are there any solutions? Flash is an analog thing. It does not even apply to ISDN. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] OT: Recommendation for Dynamic DNS on Meshbox?
Colin Anderson wrote: I'm going to do a deployment of LocustWorld MeshBoxes in some of our remote locations. Build 90 comes with Asterisk 1.0, and our plan is to use the MeshBoxes as a WAP for non-Asterisk uses but also to add a 2nd NIC to deploy Snom's in the remote location. This works fine (was suprisingly easy to do), and I have the Meshbox running Asterisk and IAX'ing to our primary no problem. They will be on a cable ISP for the broadband that only gives dynamic IP's. I would like to use a Dynamic DNS service so we can address the boxes as foo.somewhere.com for remote managment. Does anyone have a recommendation of a free / non free etc for a Dynamic DNS service? Note the Meshbox is a brutally stripped 2.4.x kernel so deploying software to it is a problem, dependency hell and I also don't want to mess too much with the base config. Ideally, a stand alone executable for the client update or a really really well crafted shell script. I can get wget and cron running on it so I am considering writing a web service that sits on our primary end and cron can call wget and POST the output of IFCONFIG; the web service can parse out the dynamic IP from the IFCONFIG output. Another possibility is to grab the IP from the wiana.org registry; unfortunately they don't have a web service for it so it would involve a script that parses out the IP from the HTML, but that would break if wiana decides to change the HTML or if wiana discontinues operations. Asterisk is VERY bad at dealing with DNS. Even a transient problem will break Asterisk. Specifically, have hosts, rather than IP addresses, in sip.conf, iax.conf, etc. Then start Asterisk with no DNS available. Just power off your NAT box. After Asterisk has started, turn your NAT router back on. Let the fun begin. I don't know of this has been fixed in CVS-HEAD or not. --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] E1/T1 back to back ??
On 07:46 PM 3/13/2005, Brett, Gary wrote: Hi there Just a quick question, I will be building some servers in a lab utilizing Digium E1 cards. I would like if possible to avoid the expense of installing an e1/ISDN30 in my lab. I have two questions really, first does anybody know of an effective simulation tool I can use to replicate a real world PRI but without the telco line being installed. And secondly, can I have a scenario with 2 asterisk servers with digium e1 cards 'back to back' one configured as the network side and the other configured as the client side (can I just use a single cat5 straight through cable between them ?? and cant the Digium e1 cards operate ok in both modes?) Any advice would be greatly appreciated Yes, you can connect two asterisk systems together back to back using t1/e1 interfaces. You will need a T1 crossover cable (do a google on T1 Crossover). Make sure you set signalling to net and cpe (if using pri signalling). ___ 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] Cisco 7960 SIP 7.4
On Mar 14, 2005, at 5:20 AM, Doug Lytle wrote: For those that are interested, I was just out on the Cisco site and noticed that they had released firmware 7.4 as of March 11th for the 7940/7960 phones. I don't see any major changes in the release notes--mostly small bug fixes. They fixed some DHCP and NTP problems, as well as a 802.1x problem with some of their switches. There were a couple SIP protocol fixes in there too, plus a spelling fix. In other words, if things are working for you right now, there's probably no reason to upgrade. Scott ___ 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] DS3 with Asterisk
Get an M13 from adtran and split it. You could also get a Cisco AS5400 Michael Blood wrote: I have done some research on the discussions that have occured on this list about DS3s with Asterisk. It seems to be dead and I have not found any active work on the project. I know that a full DS3 may have some technical limitations with why they may not work with Asterisk but I am interested in utilizing a partial DS3. Is there anyone utilizing DS3s out there with asterisk at all and if so how are you implementing it? (Splitting? Custom Drivers? Etc..) Has anybody attempted/failed/succeeded to make a DS3 to asterisk work? Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream GXP-2000
FYI, spoke with Grandstream this morning, the GXP-2000 release has been delayed again. Looking like April now before these hit the street. -- Cory Andrews Senior Partner VOIPSupply.com + V: 800.398.VOIP X22 F: 716.630.1548 E: [EMAIL PROTECTED] ___ 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] 1.0.5 / 1.0.6 and oh323 compiling problem
Hi, I have the same problem with cvs head. (1.0.6) See http://www.inaccessnetworks.com/projects/asterisk-oh323 And https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php (issue 00...008) some 'patch' files are included. I am a newbie to linux and asterisk. I do not want to blow my config. Please give me a feed-back if those files helped you and how. Also if you have a work-around (like an old file to use) Thanks regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] -Original Message- From: Dmitry Melekhov [mailto:[EMAIL PROTECTED] Sent: lundi 14 mars 2005 11:52 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 1.0.5 and h323 compiling problem Hello! Looks like h323 compiling is FAQ, but I didn't found an answer... The same problem with 0.6.5 and 0.7.1: gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/var/local/files/asterisk-1.0.5/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c (... snip ...) How can I compile h323 with Asterisk 1.0.5? Thank you! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime does not work yet, ...
INSERT INTO sip_buddies VALUES (1,'621',NULL,NULL,NULL,'\Demo\,621','yes','inhouse',NULL,'rfc2833',NULL ,N ULL,'dynamic',NULL,NULL,NULL,NULL,'[EMAIL PROTECTED]',NULL,'yes',NULL,NULL,NULL,'1', '' ,'999',NULL,NULL,NULL,'Password','friend','621','ulaw;alaw','all',NULL,0,'', 'y Try using commas instead of semicolons. I'm using commas and its working: Codecs : 0x10c (ulaw|alaw|g729) Codec Order : (g729|ulaw|alaw) -Matthew ___ 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] Panasonic KX-TD1232
On Mon, 14 Mar 2005, Jerry Geis wrote: I have connected the KX-1232 to asterisk with the T1 card. Is it dissappointing though as I have not gotten any Caller id information running over the T1. But it does function. We have callerid working with that setup (well, actually an E1). You can contact me off-list if you need any assistance. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime does not work yet, ...
Matthew Boehm wrote: INSERT INTO sip_buddies VALUES (1,'621',NULL,NULL,NULL,'\Demo\,621','yes','inhouse',NULL,'rfc2833',NULL ,N ULL,'dynamic',NULL,NULL,NULL,NULL,'[EMAIL PROTECTED]',NULL,'yes',NULL,NULL,NULL,'1', '' ,'999',NULL,NULL,NULL,'Password','friend','621','ulaw;alaw','all',NULL,0,'', 'y Try using commas instead of semicolons. I'm using commas and its working: Codecs : 0x10c (ulaw|alaw|g729) Codec Order : (g729|ulaw|alaw) I tried to use comas as well, rebooted the phone, reladed * and still the same: 601 can call 621, but 621 cannot call 601 Codecs : 0x0 (nothing) Codec Order : (none) bye Ronald ___ 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] How to flash a modem line
The H0,H1 timing may be tricky, but, If the modem is AT compliant, ATD! is
the flash command. The timing is based on S register 29.
- Original Message -
From: Raoul Bönisch [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, March 14, 2005 10:50 AM
Subject: [Asterisk-Users] How to flash a modem line
* Mateo Meier [EMAIL PROTECTED] [2005-02-28 09:13]:
I tryed that with capi.. but no luke. It will hang up the line anyway :-(
exten = s,1,Playback(transfer)
exten = s,2,Flash(capi/72044**:041720,18)
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
Any idears why ?
No wonder! The Flash() application can flash zap channels only
and you are trying to flash a modem channel. There's no obvious
facility to flash a modem channel as I can find nothing about it
in chan_modem.c. :-\
I want to do the same thing and I am using ISDN4Linux. Perhaps we
can try submit AT-commands to the modem line. A timed H0 and H1
command would do the thing. I'm afraid we'd have to change the
asterisk source code though. It would even work with CAPI as you
can use ISDN4Linux on top of CAPI, too.
Please tell me if you find any solutions :-)
Greets!
Raoul
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Re: [Asterisk-Users] Asterisk Billing System
Hi! http://www.voip-info.org/wiki-Asterisk+addon+rate-engine or write you own AGI :) (see ASTCC) Best regards, Madhawa On Fri, 11 Mar 2005 19:18:24 -, Kanishka Somaratne [EMAIL PROTECTED] wrote: Hi Is there a billing system that i can view all the call taken by SIP clients in asterisk ___ 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 e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended addressee, or the person responsible for delivering it to them, you may not copy, forward disclose or otherwise use it or any part of it in any way. To do so may be unlawful. If you receive this e-mail by mistake, please advise the sender immediately. ___ 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] not ringing when place outgoing call
I have configured asterisk to act as a B2BUA, so can use ser for sip proxying and forward the call to a sip provider. The problem now is that when I place a call to an outgoing number I don't hear nothing up to the time the callee responds. The first time I configured asterisk it was working fine. Now seems something changed but I am pretty newbiw to understand where the problem is. Can someone help me to understand what is going on? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco and Asterisk
Well, I'm just leaving demo in for testing. Once I get things working
I'll be changing all that to city names most likely.
I don't want the call to it the Cisco then redirect to the Asterisk
box. If I hit extension 602 right now, it works fine. What I'm trying
to do is dial out to another real phone number through the Cisco's
FXO ports (one of which is connected to an FXS.)
On Mon, Mar 14, 2005 at 02:02:08PM +0100, Tomasz Bukowski wrote:
Hi!
First of all , (apart from solving your problem) you really should get
rid of the whole [demo] context from extensions.conf, and place your
stuff in your own context (i.e. [local]) (just for convenience and
security). Getting back to the problem - as I see it you want to dial
out through Cisco gw by dialing 1XXX
To do so you must send the whole number to the gateway, so the gateway
could do something (anything) with it.
Your extensions.conf should be more like:
exten = _1XXX,1,Dial(SIP/[EMAIL PROTECTED])
Dialing 1602 on your system-phone will result with sending the number
1602 to the gateway, which will then (according to your current
dial-peer configuration) strip the leading 1 and send 602 back to
Asterisk to dial your laptop.
Hope it helps
Brgs
Tomek
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ben Miller
Sent: Friday, March 11, 2005 12:40 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco and Asterisk
Hey all,
I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can
get
a bit of help here.
First I'll explain my setup, and then my problem.
Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2
FXO
ports. I have an analog phone line plugged into the first port
(voice-port 1/0/0). I've got it setup so that calls coming into that
analog line are transferred to my Asterisk server via SIP.
In the second port on my FXO card, I have a phone cable plugged into a
phone-system phone (the kind you have in the office plugged into your
phone system, the extra port on it acts as an FXS so a normal phone
can
be plugged into it and can dial out by hitting 9,9 and then a number).
Incoming calls come into my * box fine, and I can hit digits on the
phone and have different thing happen. For example, I setup XLite on
my
work laptop and I've got an extension setup to dial my laptop. What
I'm
trying to do, though, is setup an extension that will connect back to
my
router and let me make an outgoing call on the second voice port.
Every
time I try to do this, I get SIP errors in the * CLI:
Got SIP response 400 Bad Request - 'Malformed/Missing URL' back from
206.222.200.46.
206.222.200.46 is the IP of my router. I'm pretty sure that I'm just
missing some config in my router, but I've been googling the past few
days and can't get anything that's helping. Thus, I turn to you to
help
me out, if possible.
I work for an ISP and what we eventually want to do is setup VoIP for
our broadband customers so they can do unlimited dialing to various
cities where we have routers, and we'll just through some voice ports
into those routers and get some lines hooked up.
Here is my relevant config:
sip.conf:
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
dtmfmode=inband
nat=never
promiscredir = yes ; If yes, allows 302 or REDIR to non-local SIP
address
[voice-gw] ; This is what I've setup for my Cisco
; has the voice ports
context=demo
type=friend
host=206.222.200.46 ; IP address of Cisco gateway
dtmfmode=inband
disallow=all
allow=ulaw
nat=no
qualify=yes
[ben] ; my work laptop
context=demo
type=friend
username=ben
host=dynamic
disallow=all
allow=ulaw
extensions.conf:
[general]
static=yes
writeprotect=no
; You can include other config files, use the #include command
(without
the ';')
; Note that this is different from the include command that includes
contexts within
; other contexts. The #include command works in all asterisk
configuration files.
;#include filename.conf
; The Globals category contains global variables that can be
referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for
Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface
for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any
Re: [Asterisk-Users] How to flash a modem line
* Stu Gotz [EMAIL PROTECTED] [2005-03-14 16:56]: The H0,H1 timing may be tricky, but, If the modem is AT compliant, ATD! is the flash command. The timing is based on S register 29. Yes, that's another possibility. We're close to it. I have the idea of using the System() application to call a program flashing the modem. This should be quite easy. :-) Raoul ___ 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] qualify and NAT....
Hello, I'm trying to run an ATA behind a NAT device, and am confused on exactly what the qualify config option does, other than send NOTIFY packets. Outbound calls work fine, but inbound calls do not go through. With qualify=yes and nat=yes, my show sip peers looks like: 777001/777001 10.0.0.10 D N 255.255.255.255 1222 OK (36 ms) So, it has established a connection with the peer on port 1222, however, when an incoming call comes in, it instead tries to go to port 5060, which doesn't work. I know I could use port forwarding, but that won't work well for multiple ATAs. So, am I right in thinking Asterisk should automagically forward the call to the port listed in show sip peers or am I missing something? Any help DEFINATELY appreciated!! Brian ___ 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] Skype - Bandwidth
Hi All, Does anyone know the amount of memory used by skype? regards César ___ 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] qualify and NAT....
Brian McCrary wrote: Hello, I'm trying to run an ATA behind a NAT device, and am confused on exactly what the qualify config option does, other than send NOTIFY packets. Outbound calls work fine, but inbound calls do not go through. With qualify=yes and nat=yes, my show sip peers looks like: 777001/777001 10.0.0.10 D N 255.255.255.255 1222 OK (36 ms) So, it has established a connection with the peer on port 1222, however, when an incoming call comes in, it instead tries to go to port 5060, which doesn't work. I know I could use port forwarding, but that won't work well for multiple ATAs. So, am I right in thinking Asterisk should automagically forward the call to the port listed in show sip peers or am I missing something? Any help DEFINATELY appreciated!! Qualify will make Asterisk send an OPTIONS packet. This allows Asterisk to see latency of the response to the OPTIONS packet (this does NOT test ICMP latency like ping does). This gives Asterisk a GENERAL idea of how lagged the device is. Since Qualify sends packets every once in a while (every 2 seconds?) it will also cause the dynamic port forwarding of your NAT router to keep the UDP translation active. You could set the registration interval for your SIP device to some really low number like 60 seconds and that will accomplish the same thing as the qualify=yes option. Remember clients send packets from a random high port number which changes. Port forwarding on your router is pretty useless. nat=yes combined with qualify=yes should cause enough traffic on the right ports to keep the NAT translations open on your NAT router. Now, if ASTERISK is behind NAT it's a whole other set of issues and fixes, but you don't mention that so I won't cover it. --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] SIP MWI and MySQL Realtime
Hi Mike: I've been searching for something like this script to solve my wmi problem. I used the script and it works fine only if the user leaving a message press the # key or when I search my voicemail I leave the Main with the # key. If me or the user (caller) leaving the message hang up then the script won't work. Is there a way to fix this? Thanks in advance, JO Mike Machado wrote: I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to send_mwi.pl and chmod 755 it. See top of file for notes on usage and configuration. If you have any feedback, let me know. ___ 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 -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk support for SIP REFER message
Hi I need to know if Asterisk supports the full features of the SIP REFER message (i.e blind and supervised transfers). I'm trying to do a supervised transfer through Asterisk from a VoiceXML application using the transfer tag and setting bridge=true (i.e transfer name=transfer1 bridge=true connecttimeout=10s ) but as soon as Asterisk receives the SIP REFER message generated by the VoiceXML application, it sends back a NOTIFY message with a subscription-state:terminated as if it was a blind transfer (bridge=false) which instructs the VoiceXML application to disconnect so it no longer supervises the call to get back the result ( callee unavailable, busy,...) . Usually, when the brige=true is set in the VoiceXML application, the end point that receives the SIP REFER should send a NOTIFY message with subscription-state:active and then it should send back NOTIFY messages to tell the VoiceXML application about the result of the call (i.e callee unavailable, busy,...). Regards, Gilbert Abboud M.Eng. Computer Engineering Programmer Analyst Excendia, Montreal ESN: 514-765-8490 ___ 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] TDM400 audio problems
Sorry everyone, I know this has been hashed over a bunch of times but I can't find anything that pertains to specific cracking and popping on the FXO modules of a TDM04. This happens on inbound or outbound calls. This is the first install I have done with a TDM card for FXO modules so please, be kind if I am missing something really simple. Damn I wish everyone wanted t-1's or pri's! Everything sounds great UA to UA and the dialplan works but any calls through the TDM card sound like crap. Getting bunch of crackling and ticking in the audio. I swapped the tdm card with a spare x100 clone that I had lying around and the audio is fine. This is a p4 3.4 proc with 4 gig of DDR2 on an Intel server board. zaptel.conf loadzone = us defaultzone=us #fxsks=1-4 fxsks=1 zapata.conf [channels] language=en context=incoming_grace signalling=fxs_ks usecallerid=yes echotraining=yes group=1 immediate=yes ;channel=1-3 channel=1 ;group=2 ;context=incoming_institute ;signalling=fxs_ks ;usecallerid=yes ;echotraining=yes ;immediate=yes ;channel=4 Thanks, Jason Kawakami www.optellabs.com Salt Lake City, UT ___ 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] TDM400 audio problems
Sorry everyone, I know this has been hashed over a bunch of times but I can't find anything that pertains to specific cracking and popping on the FXO modules of a TDM04. This happens on inbound or outbound calls. This is the first install I have done with a TDM card for FXO modules so please, be kind if I am missing something really simple. Damn I wish everyone wanted t-1's or pri's! Everything sounds great UA to UA and the dialplan works but any calls through the TDM card sound like crap. Getting bunch of crackling and ticking in the audio. I swapped the tdm card with a spare x100 clone that I had lying around and the audio is fine. This is a p4 3.4 proc with 4 gig of DDR2 on an Intel server board. Usaually those types of problems are associated with motherboard interrupt issues. Might try analyzing the following commands: cat /proc/interrupts (check to see if wctdm is sharing an interrupt with something else. If so, try to move the tdm card to another pci slot, or, look in your bios setup to disable unused interrupts.) run 'zttool' to see if you have any irq misses. If so, those need to be resolved. ___ 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] SIP MWI and MySQL Realtime
FYI, you can stop using that script and start using the RealTime cache ability. -Matthew Jose R. Ortiz Ubarri wrote: Hi Mike: I've been searching for something like this script to solve my wmi problem. I used the script and it works fine only if the user leaving a message press the # key or when I search my voicemail I leave the Main with the # key. If me or the user (caller) leaving the message hang up then the script won't work. Is there a way to fix this? Thanks in advance, JO Mike Machado wrote: I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to send_mwi.pl and chmod 755 it. See top of file for notes on usage and configuration. If you have any feedback, let me know. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting NAT=yes for not NATed clients
Hello, I wonder if I would have to sacrifice anything if I set NAT=yes for all sip clients I have, regardless of whether they are behind the NAT or not. The idea is to have the setting that works regardless of whether the user is behind the NAT or not, since I'm not sure what connection that particular user has to Internet at the moment. I've tried and NAT=yes works even for those clients that are not behind the NAT. Is there any peformance problems/etc if I set NAT=yes for all clients? Thanks, Roman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Montag, 14. März 2005 18:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM400 audio problems Sorry everyone, I know this has been hashed over a bunch of times but I can't find anything that pertains to specific cracking and popping on the FXO modules of a TDM04. This happens on inbound or outbound calls. This is the first install I have done with a TDM card for FXO modules so please, be kind if I am missing something really simple. Damn I wish everyone wanted t-1's or pri's! Everything sounds great UA to UA and the dialplan works but any calls through the TDM card sound like crap. Getting bunch of crackling and ticking in the audio. I swapped the tdm card with a spare x100 clone that I had lying around and the audio is fine. This is a p4 3.4 proc with 4 gig of DDR2 on an Intel server board. Usaually those types of problems are associated with motherboard interrupt issues. Might try analyzing the following commands: cat /proc/interrupts (check to see if wctdm is sharing an interrupt with something else. If so, try to move the tdm card to another pci slot, or, look in your bios setup to disable unused interrupts.) run 'zttool' to see if you have any irq misses. If so, those need to be resolved. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Log Error
That was already with SET VERBOSE 255. /rg On Mar 14, 2005, at 3:32 AM, Tzafrir Cohen wrote: On Sun, Mar 13, 2005 at 07:22:58PM -0600, Anton Krall wrote: So far nobody has answered this post... Anybody has seen this error before? Could you use a more verbose logging? IIRC, the technology is the channel type, e.g: sip, zap, iax. Somewhere something is getting either an empty channel name or some garbage as the channel name. [trimmed bottom-posting] ___ 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] Setting NAT=yes for not NATed clients
Is there any peformance problems/etc if I set NAT=yes for all clients? nat=yes causes Asterisk to respond to the *public* source port and IP address. Therefore, the only time you should ever have a problem is when the packets should not go to that port/address, which I think is close to never. Nabeel ___ 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] Setting NAT=yes for not NATed clients
Thanks! Could that mean any security problems? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Montag, 14. März 2005 19:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Setting NAT=yes for not NATed clients Is there any peformance problems/etc if I set NAT=yes for all clients? nat=yes causes Asterisk to respond to the *public* source port and IP address. Therefore, the only time you should ever have a problem is when the packets should not go to that port/address, which I think is close to never. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP MWI and MySQL Realtime
Matthew Boehm wrote: FYI, you can stop using that script and start using the RealTime cache ability. Thanks for the hint. But where can I find the cache information? I search at www.voip-info.org and couldn't find anything. Thanks! JO -Matthew Jose R. Ortiz Ubarri wrote: Hi Mike: I've been searching for something like this script to solve my wmi problem. I used the script and it works fine only if the user leaving a message press the # key or when I search my voicemail I leave the Main with the # key. If me or the user (caller) leaving the message hang up then the script won't work. Is there a way to fix this? Thanks in advance, JO Mike Machado wrote: I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to send_mwi.pl and chmod 755 it. See top of file for notes on usage and configuration. If you have any feedback, let me know. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ 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] Setting NAT=yes for not NATed clients
Roman Zhovtulya wrote: Hello, I wonder if I would have to sacrifice anything if I set NAT=yes for all sip clients I have, regardless of whether they are behind the NAT or not. The idea is to have the setting that works regardless of whether the user is behind the NAT or not, since I'm not sure what connection that particular user has to Internet at the moment. I've tried and NAT=yes works even for those clients that are not behind the NAT. Is there any peformance problems/etc if I set NAT=yes for all clients? A few SIP devices won't work with nat=yes. The Uniden UIP200 comes to mind. I don't know of any others. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] School design question
My school district will be building a new elementary school in 2006. We were about to go to bid with a traditional intercom system for the campus but I would like implement Asterisk at the campus. My question is, do we build in a traditional intercom/paging system and tie that into the Asterisk PBX, the way such intercoms have been connected to other PBX's in our district in the past, or do we put IP phones in the classrooms and tie that into a PA system for paging? Are there IP based paging systems that could be used instead of the traditional PA/loudspeaker systems in most schools? We will be writing a spec shortly (and I will be seeking a consultant on asterisk-biz soon to assist us), but I need to know whether I need to get our engineering consultant to redraw the cabling to reflect a data jack at the location of the classroom phone instead of the telco jack that is currently on the plans. Thanks in advance for the advice! -- Chris Hobbs Silver Valley Unified School District Head geek: Technology Services Coordinator webmaster: http://www.silvervalley.k12.ca.us/~chobbs/ postmaster: [EMAIL PROTECTED] pgp: http://www.silvervalley.k12.ca.us/~chobbs/key.asc ___ 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] Asterisk@Home
I think [EMAIL PROTECTED] will work well for what you want to do. The GUI also allows you to edit the config files. it just saves time it dosn't reduce the functionality of Asterisk. if you get the IM callbacks feature working I would be interested. This would be a great feature to include in [EMAIL PROTECTED] --- Scheda [EMAIL PROTECTED] wrote: Have any of you tried this? http://asteriskathome.sourceforge.net/ I'm thinking of using this version. I'm debating between it and Knoppix with Asterisk thrown in there as well. I'm a linux newbie for the most part, but can get around and get done what I need done with help here and there, but I don't know if [EMAIL PROTECTED] is all what I need. Here is what I need * to do for me pretty much. -Voicemail -Conferencing -IM callbacks (Instant message from my cell and it calls me back) -Extentions -A few other things which I can handle I've used * a tad bit in the past, no real heavy work with it though. I would think that the GUI in [EMAIL PROTECTED] would make Asterisk less functional. Seeing as how I'm not using this for some large business or anything, just as something so listeners of a radio show I do can interact, I don't think [EMAIL PROTECTED] would be WAY too restrictive on what I need accomplished. If you have used this before, can you post a review on what you think of it or just tell me if this will suit my needs or not? ___ 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 __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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] SIP MWI and MySQL Realtime
You can find the cache information in the sip.conf inside /usr/src/asterisk/configs/ (or whereever you keep your source). -Matthew Jose R. Ortiz Ubarri wrote: Matthew Boehm wrote: FYI, you can stop using that script and start using the RealTime cache ability. Thanks for the hint. But where can I find the cache information? I search at www.voip-info.org and couldn't find anything. Thanks! JO -Matthew Jose R. Ortiz Ubarri wrote: Hi Mike: I've been searching for something like this script to solve my wmi problem. I used the script and it works fine only if the user leaving a message press the # key or when I search my voicemail I leave the Main with the # key. If me or the user (caller) leaving the message hang up then the script won't work. Is there a way to fix this? Thanks in advance, JO Mike Machado wrote: I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to send_mwi.pl and chmod 755 it. See top of file for notes on usage and configuration. If you have any feedback, let me know. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Has anybody tried NVFaxDetect Fax detection SIP/IAX
Has anybody tried NVFaxDetect Fax detection for sip SIP/IAX channel? There is a new application from Newman Telecom for fax detection. http://www.sineapps.com/news.php?rssid=575 Current Asterisk Fax detection doesn't work for me as I don't have Digium cards; I'm using Siupra -- #Joseph ___ 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] Skype - Bandwidth
On Mon, 2005-03-14 at 14:30 -0300, César Davi Ávila do Nascimento wrote: Hi All, Does anyone know the amount of memory used by skype? Did you think about the best venue to ask this question. We are not a skype support forum. And BTW, TURN OFF HTML EMAIL. Exercise some thought about those who will read your message before assuming we all need to see blue. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] School design question
On Mon, 2005-03-14 at 10:46 -0800, Chris Hobbs wrote: My school district will be building a new elementary school in 2006. We were about to go to bid with a traditional intercom system for the campus but I would like implement Asterisk at the campus. My question is, do we build in a traditional intercom/paging system and tie that into the Asterisk PBX, the way such intercoms have been connected to other PBX's in our district in the past, or do we put IP phones in the classrooms and tie that into a PA system for paging? Are there IP based paging systems that could be used instead of the traditional PA/loudspeaker systems in most schools? You may want to continue having a general access PA that is just like your currently installed system. Your PA probably needs to be easily understood in the hallways as well as the classrooms. Cost of wire and speakers are much lower than an IP phone and the extra power requirements and possibly buggy firmware. We will be writing a spec shortly (and I will be seeking a consultant on asterisk-biz soon to assist us), but I need to know whether I need to get our engineering consultant to redraw the cabling to reflect a data jack at the location of the classroom phone instead of the telco jack that is currently on the plans. If you aren't planning on data to the classroom, you probably are already behind. You probably should plan on running 2 cat5 cables to every room. At worse, you use 1 cat5 for plain old telecom. You at least have options at that point. See about running them all to nice patch panels so that you just make jumpers from the kind of network you want over to the port that needs it. Do consider that you don't have to purchase fancy phones for the classrooms. You could use analog telephones that are cheap to replace and use a group of channel banks to support the phones. Maybe a bit more expensive than the IP phones, but it is tried and proven technology. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] colinux fresh install, zaptel does not compile, size_t error
Hi, Thanks Andreas ! On my colinux I add to dig into the colinux source code to extract the .config file (it was missing from /root on my install). After that, I did make-kpkg as explained in your page. Then, the compilation error disappeared. The whole issue was about some missing kernel include files. Not only do you need to get the kernel source... you also need to build some of it to get these missing files. Many thanks again. Yours, JeanHuguesRobert At 14:48 14/03/2005 +0100, you wrote: [EMAIL PROTECTED] wrote: I followed the instructions on http://www.asterisk.org/index.php?menu=download. I picked the latest version using CVS. Things went fine until I cd zaptel ; make clean ; make install. I then get an error when compiling zaptel.c /usr/src/linux/include/linux/kernel.h:75: error: parse error before size_t This happens very early on and I suspect that it is actually an issue with the kernel include files on my machine. Nota: I am installing on a colinux debian. uname -a Linux colinux2 2.4.26-co-0.6.1 #1 Sat May 29 15:30:37 IDT 2004 i686 GNU/Linux On http://www.ramdyne.nl/ you can find an article on how I got rid of the same problems you were having (on a Debian sarge install). Unfortunately the server is down for the next couple of hours... Here's a link to the google cache copy: http://66.102.9.104/search?q=cache:pR1IMCaiRcQJ:www.ramdyne.nl/index.php%3Fcat%3D11+%2Basterisk+%2Bramdyne+%2Bdebianhl=nlstart=1 -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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 - Web: http://hdl.handle.net/1030.37/1.1 Phone: +33 (0) 4 92 27 74 17 ___ 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
