[asterisk-users] enable eyeBeam to accept only one call
Hello I'm using eyeBeam, and Asterisk keeps sending my clients a second call, when they are still in one call (because eyeBeam has lots of channels). I was using X-Lite (with 3 channels) and Asterisk never sent the client a second call. How can I force Asterisk (or eyeBeam) just to send one call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao Pereira -- __ João Gomes Pereira FCCN Av. do Brasil, nº 101 1700-066 Lisboa tel: +351 218 440 100 - fax: +351 218 472 167 email|SIP: [EMAIL PROTECTED] http://www.fccn.pt __ --- Aviso de Confidencialidade Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha recepcionado indevidamente esta mensagem, solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218 440 100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218 440 100 and delete it immediately. --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple PBXs in one box
Hello I would like to know if it is possible to have multiple PBXs implemented in one Asterisk box. I have different companies using my Asterisk server (remotely) and I don't want them to be calling each other. I want to create different profiles in which my clients can only see its own PBX. Each PBX will have its extensions and outbound/inbound routes... but everything in only one Asterisk. Is this possible? How can I implement it? Creating different contexts? Should I use a special software together with Asterisk? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dial-out call queue
Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk crash and debug
Hello each 15 days my Asterisk crashes. Every time it happens I try to change something in its configuration to avoid the next crash. I already checked the logs but I don't know what to do. Can someone tell me whats the problem? These are my Asterisk logs: http://vox.fccn.pt/crash Thanks Regards Joao Pereira ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center SoftPhone with Auto Answer
I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. I'm using X-Lite, and in X-Lite I can't force the users to answer the call. The users can put Auto Answer = Off. Also, the response from Counterpath was weird, as they said they're engineering team cannot remove the Auto Answer option: To have the auto-answer permanently on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu So, if someone knows a nice softphone for an Asterisk Call Center, please advice me. Thanks Regards Joao Pereira Ed Pastore wrote: On Sep 17, 2007, at 11:11 AM, Joao Pereira wrote: But still, the user can choose not to answer the phone. I want to force the users to accept the calls. Wouldn't that be the same as paging/intercom, then? http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center SoftPhone with Auto Answer
But still, the user can choose not to answer the phone. I want to force the users to accept the calls. Regards Joao Pereira Thiago Maluf wrote: Ola Joao, tem um modo do Asterisk fazer isso sim. Entre em contato no meu GTALK por esse e-mail e eu te dou mais informações. Abs! Hi List, The asterisk have one way to do it. just put one script to discovery if this user is online or offline. case is offline play one music. if not, call the user. understand? thiago! 2007/8/6, Joao Pereira [EMAIL PROTECTED]: Hello I need a Softphone with auto answer where users can't turn it off. Does someone knows a softphone where users can't turn the auto answer off? Or is there any way Asterisk could force the clients to answer the phone? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Center SoftPhone with Auto Answer
Hello I need a Softphone with auto answer where users can't turn it off. Does someone knows a softphone where users can't turn the auto answer off? Or is there any way Asterisk could force the clients to answer the phone? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk SIP domain (in LAN or DMZ)?
Hello I want to use Asterisk to implement a SIP Domain allowing my clients to do URI dialing and receive calls from the Internet through URIs and ENUM. My question is, should I put my Asterisk outside the firewall (in the DMZ) to allow connections to the Internet? Or should I have it inside my local network and put a SIP Proxy (like Openser) in the DMZ to implement the SIP domain? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] compile problem with wavelenght
Hello Im trying to install an old version of Asterisk. But it isnt working: when I run make install: gcc -o gentone gentone.c -lm ./gentone busy 480 620 Wavelength 1 (in samples): 16.7 Minimum samples (1): 50 (3.00.3 wavelengths) Wavelength 1 (in samples): 12.90323 Minimum samples (1): 400 (31.00.3 wavelengths) Need 400 samples Wrote busy.h ./gentone ringtone 440 480 Wavelength 1 (in samples): 18.18182 Minimum samples (1): 200 (11.00.3 wavelengths) Wavelength 1 (in samples): 16.7 Minimum samples (1): 50 (3.00.3 wavelengths) Need 200 samples Wrote ringtone.h gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -fPIC-c -o chan_oss.o chan_oss.c gcc -shared -Xlinker -x -o chan_oss.so chan_oss.o -ldl -lpthread -lncurses -lm -lresolv -lssl gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -fPIC-c -o chan_phone.o chan_phone.c chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/services/asterisk/asterisk-1.2.10/channels' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk-1.2.10]# Whats happening? I already tried with 3 different versions downloaded from asterisk.org site. Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compile problem with wavelenght
Hello Thanks a lot for the help. I just commented these lines and its working: #ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/ixjuser.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/ixjuser.h),) # CHANNEL_LIBS+=chan_phone.so #endif I just hope that this doesnt bring me problems in the future :P Thanks regards Joao Pereira Tzafrir Cohen wrote: On Thu, Apr 12, 2007 at 10:25:37AM +0100, Joao Pereira wrote: Hello Im trying to install an old version of Asterisk. But it isnt working: when I run make install: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -fPIC-c -o chan_phone.o chan_phone.c chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/services/asterisk/asterisk-1.2.10/channels' make: *** [subdirs] Error 1 This is a known problem that has been fixed in later versions of asterisk 1.2 . Alternatively, build the same version withough building chan_phone.so . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum retries exceeded on transmission
Hello Thanks a lot for your reply. Im now using asterisk-1.2.10 and the problem disappeared. Thanks regards Joao Pereira Edoardo Serra wrote: Same to me !! Calls from OpenSER to Asterisk It happens only with Asterisk versions = 1.2.14 I'm going to capture some traffic Tnx for help Regards Alex Balashov ha scritto: Joao, It sounds like the proxy is not acknowledging the Asterisk's processing of the INVITE, but I could be wrong. It would be helpful to supply a packet capture between OpenSER and Asterisk so we could see the setup flow. Thanks, -- Alex On Tue, 10 Apr 2007, Joao Pereira said something to this effect: Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maximum retries exceeded on transmission
Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no reply to our critical packet
Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
I think it can be done, but not with a GrandStream HandyTone ATA because the manual says this: What it CANNOT do: - Terminate a VoIP call into the PSTN port - Allow a call from PSTN to route other VoIP devices (different from the FXS phone) over the IP network - Automatically route calls made by the local user to PSTN line so, if it cant terminate VoIP calls into the PSTN, it cant forward VoIP calls to the Dock and Talk. Joao Dovid B wrote: There has been talk about it before and I think people have done it. Paging Sam Tam - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Tuesday, January 02, 2007 4:56 PM Subject: Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source Do you know If its possible to do the same with Dock and Talk and an ATA GrandStream HandyTone 386? Thanks Joao Pereira Jonathan Attwood wrote: I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk. Because I'm using Asterisk, I cannot use voice dialling, however inbound outbound calls work extremely well. I have Asterisk outbound routes set up to make a calls to cell phones go through the Dock-n-Talk. On 1/1/06, *Brian McEntire* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Is anyone familiar with cell phone switches that allow routing cell phone calls through in-home wiring? One example of these devices is the Phone Labs Dock-N-Talk. It says it keeps your cell charged when you are home and connects your cell (for incoming and outgoing calls) to your home wiring or cordless phones. But it also has features such as allowing speed dialing and voice dialing from extensions if your cell phone has those features. So I'm not sure if the device offers a fully compatible FXO signalling. I'm currently running Asterisk with 1 POTS and 1 VOIP (via Sipura 3000) lines coming into Zaptel FXS modules, and then I have two FXO modules for two extensions. I'm thinking of doing away with the land line. Should something like the Dock-N-Talk allow substituting a cell phone line for the POTS line? ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM loses server registration
Hello to all When my SNOM (300 or 320) loses Internet connectivity, it loses its Asterisk registration (ok, thats normal). But when the phone is back online, he doesn't try to register in Asterisk. I believe this happens to avoid flooding the private LANs when the Internet link is lost but the problem is that the phones don't try to re-register in the future Sometimes it stays 2 hours without registering to Asterisk. When this happens, the only solution is to reboot it (and hear the users complains) :( How can I avoid this? How can I reduce the time to re-register in SNOM 300 or 320 ? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
Do you know If its possible to do the same with Dock and Talk and an ATA GrandStream HandyTone 386? Thanks Joao Pereira Jonathan Attwood wrote: I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk. Because I'm using Asterisk, I cannot use voice dialling, however inbound outbound calls work extremely well. I have Asterisk outbound routes set up to make a calls to cell phones go through the Dock-n-Talk. On 1/1/06, *Brian McEntire* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Is anyone familiar with cell phone switches that allow routing cell phone calls through in-home wiring? One example of these devices is the Phone Labs Dock-N-Talk. It says it keeps your cell charged when you are home and connects your cell (for incoming and outgoing calls) to your home wiring or cordless phones. But it also has features such as allowing speed dialing and voice dialing from extensions if your cell phone has those features. So I'm not sure if the device offers a fully compatible FXO signalling. I'm currently running Asterisk with 1 POTS and 1 VOIP (via Sipura 3000) lines coming into Zaptel FXS modules, and then I have two FXO modules for two extensions. I'm thinking of doing away with the land line. Should something like the Dock-N-Talk allow substituting a cell phone line for the POTS line? ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens Gigaset SL75
Do you know if it has 802.1x authentication as it is defined in EDUroam ( http://www.eduroam.org/ ) ? I never found a WiFi phone working with 802.1x I tested ZyXel Prestige 2000 but the sound was bad and it doesnt support 802.1x :( Thanks Joao Pereira [EMAIL PROTECTED] wrote: No, the Gigaset is the only WLAN phone I tested so long, so I can not compare it to the other phones you mentioned. -Original Message- *From:* Olivier [mailto:[EMAIL PROTECTED] *Sent:* Friday, November 24, 2006 10:19 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] Siemens Gigaset SL75 Have you ever compared it to Linksys WIP 330 or Zyxel 2000 ? Those 2 seem to get average reviews from users (short range, autonomy, ...). On paper, it seems to me a decent WiFi phone is still lacking today. Maybe this Gigaset SL75 could fill the void. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] matching the beginning of an EXTEN
Hello how can I distinguish all the calls that arrive to my Asterisk starting with: 351217588XXX ? I want match the first 9 digits does Asterisk has any function for this? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI to SIP
For PRI you have 3 main solutions. This is the order of stability (and pricing): 1. Digium or Sangoma cards use the computer processor and that could be bad if you have huge traffic through the PRI 2. Eicon Diva cards have their own processor, which releases the PC processor and gives more stability 3. You can also use a dedicated router (ex: Cisco) to do that.Its more expensive, but more reliable. Regards Joao Pereira Patrick Fortin wrote: Hi Can someone recommend a PRI to SIP Box that work well with asterisk We are presently testing with a Patton Smartnode 2400 but we are unable to fax through it. We don't want to use digium card in a linux box for the PRI connection. Which Cisco box would work. Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] matching the beginning of an EXTEN
perfect!!!
its now working this way:
exten = _.,4,GotoIf($[ ${EXTEN:0:9} = 351217588] ? 20:10)
Thanks a lot
Joao Pereira
Ove Aursand wrote:
Use ${EXTEN:0:9}
Regards,
Ove
Joao Pereira wrote:
Hello
how can I distinguish all the calls that arrive to my Asterisk
starting with: 351217588XXX ?
I want match the first 9 digits does Asterisk has any function
for this?
Thanks
Regards
Joao Pereira
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Re: [asterisk-users] how to define a secure trunk
Can I do the encrypted trunk in SIP? Does Asterisk supports it? Thanks Joao Pereira Pavel Jezek wrote: http://www.voip-info.org/wiki/view/IAX+encryption Joao Pereira wrote: Hello I would like to define a trunk from my Asterisk to a VoIP provider, but I want to make it secure, because its through the Internet. I want to be sure no one makes calls as being me, and that my calls aren't intercepted. Is it possible to define encrypted trunks? And should I define the trunk in SIP, IAX or something else? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to define a secure trunk
Hello I would like to define a trunk from my Asterisk to a VoIP provider, but I want to make it secure, because its through the Internet. I want to be sure no one makes calls as being me, and that my calls aren't intercepted. Is it possible to define encrypted trunks? And should I define the trunk in SIP, IAX or something else? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom or Cisco Phones?
Hello I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom and Cisco Phones. Can you gurus, please, give me your impression of these 2 brands? I need to focus more in SIP and Asterisk compatibility and less in pricing (yes, I know the Cisco are more expensive). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] defining trunks in sip.conf
I just upgraded an old Asterisk 1.0.xx to 1.2 but there are some changes in the trunk definitions of sip.conf All my trunks stopped working. Is the sintax someting like this? register=200:1000:[EMAIL PROTECTED]:5060/200 this is to user 200 (why do we need to put it 3 times???) with password 1000 and to register in domain.pt I already saw the manuals but the trunks arent still working :( Can someone help me? Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client with video???
The problems with X-Lite 3 are: - just accepts one SIP registration - doesnt send video to other X-Lite or eyeBeam versions - sometimes loses the SIP informations when you reboot the PC . Regards Joao Pereira Blake Krone wrote: What's wrong with X-Lite 3.0? I haven't had any issues with it and find it to be one of the best SIP video software choices, and it's free. On 7/27/06, *Joao Pereira * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software. Does someone knows about a better software? Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk video support
Hello to all I used SER for SIP calls with video, but now Im trying the same in Asterisk and It doesnt work. I m using X-Lite 3.0 (the same that worked with SER). Do Asterisk needs any special configuration to allow SIP calls with video between its clients? Regards Joao Pereira Asterisk's support for video over SIP is very rudimentary. Only to video codecs H.261, H.263, and H.263+ are supported, and even then, not very well. There is no support for dynamic negotiation of frame rates, etc. Queries to the -dev list, as to progress on these features were recently met with silence. We will be looking to jump into the project to support our own initiatives in the area of video in a few weeks. Until things change, your best bet for connecting SIP video phones is SER. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP client with video???
Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software. Does someone knows about a better software? Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] planet VIP 152 T
Hello to all Im testing a Planet 152 T phone and Im having some problems. Can someone tell me if this phone does URI dialing? And does it work behind NAT (does it need any special configuration on the SIP server)? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] regexp issue
Hello to all
I had Asterisk dialing the PSTN through a defined trunk.
But when I enabled the SIP URI calls Asterisk stopped contacting
the PSTN trunk
The SIP URI dial code (who created the problem) is this:
exten = _.,1,NoOp(Incoming Call from from-internal-custom extension
${CALLERID} for [EMAIL PROTECTED])
exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _.,6,NoOp(@${SIPDOMAIN} is remote, forwarding...)
exten = _.,7,Macro(uridial,[EMAIL PROTECTED])
exten = _.,8,HangUp()
exten = _.,10,Goto(custom-noturi,${EXTEN},1)
exten = h,1,HangUp()
How can I say that this code is just for calls to foreign domains?
Something like:if (SIPDOMAIN != fccn.pt)
Regards
Joao Pereira
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Re: [Asterisk-Users] using a billing system
yes, a2billing.php is in agi-bin: [EMAIL PROTECTED] locate a2billing.php /usr/src/a2billing/Chameleon/A2Billing_AGI/a2billing.php /var/lib/asterisk/agi-bin/a2billing.php Could be because of the missing pcntl php extension? [EMAIL PROTECTED] rpm -qa | grep php php-mysql-4.3.9-3 php-ldap-4.3.9-3 php-odbc-4.3.9-3 php-pgsql-4.3.9-3 php-4.3.9-3 php-pear-4.3.9-3 Thanks Joao Pereira Vahan Yerkanian wrote: exten = _2,1,Answer exten = _2,2,Wait,2 exten = _2,3,DeadAGI, a2billing.php exten = _2,4,Wait,2 exten = _2,5,Hangup I tried it and the call is answered bu Asterisk and never dials the destination. :( Yes that's the correct way to launch A2B script. Are you a2billing.php is in your agi-bin directory? Also, you can see if the script runs without error by executing it from shell(you'll need php cli compiled and installed) and keep pressing enter key to see the script output. Perhaps you have your php binary in the wrong path or a missing php extension. Make sure you have pcntl php extension installed too. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using a billing system
I already installed pcntl but the billing isnt workin. I followed the Asterisk2Billing wiki and putted this line in the end of sip.conf: #include additional_a2billing_sip.conf but when I dial, Asterisk answers 407 Proxy Authentication Required If I do comment the line in sip.conf ( ; #include additional_a2billing_sip.conf ) some errors appear in the Asterisk CLI: a2billing.php: [ANSWER CALL] a2billing.php: Requesting DTMF :: Len-10 May 30 19:14:51 WARNING[27515]: file.c:475 ast_openstream: File prepaid-enter-pin-number does not exist in any format May 30 19:14:51 WARNING[27515]: file.c:787 ast_streamfile: Unable to open prepaid-enter-pin-number (format gsm): No such file or directory a2billing.php: RES DTMF : -1 a2billing.php: CARDNUMBER :: -1 a2billing.php: PREPAID-INVALID-DIGITS May 30 19:14:51 WARNING[27515]: file.c:475 ast_openstream: File prepaid-invalid-digits does not exist in any format a2billing.php: PREPAID-INVALID-DIGITS a2billing.php: Requesting DTMF :: Len-10 May 30 19:14:51 WARNING[27515]: file.c:475 ast_openstream: File prepaid-enter-pin-number does not exist in any format May 30 19:14:51 WARNING[27515]: file.c:787 ast_streamfile: Unable to open prepaid-enter-pin-number (format gsm): No such file or directory a2billing.php: RES DTMF : -1 a2billing.php: CARDNUMBER :: -1 a2billing.php: PREPAID-INVALID-DIGITS May 30 19:14:51 WARNING[27515]: file.c:475 ast_openstream: File prepaid-invalid-digits does not exist in any format a2billing.php: PREPAID-INVALID-DIGITS a2billing.php: Requesting DTMF :: Len-10 May 30 19:14:51 WARNING[27515]: file.c:475 ast_openstream: File prepaid-enter-pin-number does not exist in any format May 30 19:14:51 WARNING[27515]: file.c:787 ast_streamfile: Unable to open prepaid-enter-pin-number (format gsm): No such file or directory a2billing.php: RES DTMF : -1 a2billing.php: CARDNUMBER :: -1 a2billing.php: PREPAID-INVALID-DIGITS -- AGI Script a2billing.php completed, returning 0 I already have voicemail working, so the problem is not in the audio files reader. The prepaid-enter-pin-numbers and prepaid-invalid-digits are already in the /var/lib/asterisk/mohmp3/acc_* dirs Can you give me a help to understand whats the problem? Thanks Joao Pereira Vahan Yerkanian wrote: Greetings, pcntl is a required module for a2billing. It is vital for ensuring the call is registered if it's terminated/hungup not by normal needs. What is the output from when you execute /var/lib/asterisk/agi-bin/a2billing.php? If it's saying not found, then you need to edit the php binary location in the 1st line of it. Otherwise, pressing enter continuously after running the a2billing.php from command line should start giving you the debug info. HTH, Vahan Joao Pereira wrote: yes, a2billing.php is in agi-bin: [EMAIL PROTECTED] locate a2billing.php /usr/src/a2billing/Chameleon/A2Billing_AGI/a2billing.php /var/lib/asterisk/agi-bin/a2billing.php Could be because of the missing pcntl php extension? [EMAIL PROTECTED] rpm -qa | grep php php-mysql-4.3.9-3 php-ldap-4.3.9-3 php-odbc-4.3.9-3 php-pgsql-4.3.9-3 php-4.3.9-3 php-pear-4.3.9-3 Thanks Joao Pereira Vahan Yerkanian wrote: exten = _2,1,Answer exten = _2,2,Wait,2 exten = _2,3,DeadAGI, a2billing.php exten = _2,4,Wait,2 exten = _2,5,Hangup I tried it and the call is answered bu Asterisk and never dials the destination. :( Yes that's the correct way to launch A2B script. Are you a2billing.php is in your agi-bin directory? Also, you can see if the script runs without error by executing it from shell(you'll need php cli compiled and installed) and keep pressing enter key to see the script output. Perhaps you have your php binary in the wrong path or a missing php extension. Make sure you have pcntl php extension installed too. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using a billing system
I think Asterisk2Billing is trying to play some audio file to make the callers put a PIN number. But can I use it without the PIN, and configure Asterisk2billing to check the database to see if the user exists? Thanks Joao Pereira Vahan Yerkanian wrote: Greetings, pcntl is a required module for a2billing. It is vital for ensuring the call is registered if it's terminated/hungup not by normal needs. What is the output from when you execute /var/lib/asterisk/agi-bin/a2billing.php? If it's saying not found, then you need to edit the php binary location in the 1st line of it. Otherwise, pressing enter continuously after running the a2billing.php from command line should start giving you the debug info. HTH, Vahan Joao Pereira wrote: yes, a2billing.php is in agi-bin: [EMAIL PROTECTED] locate a2billing.php /usr/src/a2billing/Chameleon/A2Billing_AGI/a2billing.php /var/lib/asterisk/agi-bin/a2billing.php Could be because of the missing pcntl php extension? [EMAIL PROTECTED] rpm -qa | grep php php-mysql-4.3.9-3 php-ldap-4.3.9-3 php-odbc-4.3.9-3 php-pgsql-4.3.9-3 php-4.3.9-3 php-pear-4.3.9-3 Thanks Joao Pereira Vahan Yerkanian wrote: exten = _2,1,Answer exten = _2,2,Wait,2 exten = _2,3,DeadAGI, a2billing.php exten = _2,4,Wait,2 exten = _2,5,Hangup I tried it and the call is answered bu Asterisk and never dials the destination. :( Yes that's the correct way to launch A2B script. Are you a2billing.php is in your agi-bin directory? Also, you can see if the script runs without error by executing it from shell(you'll need php cli compiled and installed) and keep pressing enter key to see the script output. Perhaps you have your php binary in the wrong path or a missing php extension. Make sure you have pcntl php extension installed too. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to strip a digit
you need to put :1 next to ${EXTEN}
something like:
exten = _91NXXNXX,1,AGI(call_log.agi,${EXTEN:1})
exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
exten = _91NXXNXX,3,Hangup
Joao Pereira
Erick Perez wrote:
I have the following extension to dial outside via SIP
it's like this:
phoneasterisk-internet-SIP providerUSA
exten = _91NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN},55,o)
exten = _91NXXNXX,3,Hangup
I want to strip the digit 9 before sending it to the SIP provider.
Also, any suggestions for the above definition?
Thanks,
Erick.
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Re: [Asterisk-Users] using a billing system
Its almost done but now Im failing to authenticate with my Telco's gateway, because the registration information is in sip.conf: [my-telco] type=friend host=mytelco.com disallow=all allow=ulaw allow=alaw username=username fromuser=username secret=password and I used to make SIP calls like this: exten = _,1,Dial(SIP/[EMAIL PROTECTED]) Now, with the code you gave me, Asterisk is consulting Asterisk2Billing: exten = _,1,Answer exten = _,2,Wait,2 exten = _,3,DeadAGI,a2billing.php exten = _,4,Wait,2 exten = _,5,Hangup but when I place the call, he fails to authenticate with my-telco :( How can I use the registration information that is in sip.conf and continue to use Asterisk2Billing ? Thanks Joao Pereira William Piper wrote: You need to specify which context to use in the a2billing.conf. Your extensions.conf should look like this: exten = _2.,1,Answer exten = _2.,2,Wait,2 exten = _2.,3,DeadAGI(a2billing.php|2) exten = _2.,4,Wait,2 exten = _2.,5,Hangup Also, check out http://forum.asterisk2billing.org/ for more help. bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Tuesday, May 30, 2006 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] using a billing system yes, a2billing.php is in agi-bin: [EMAIL PROTECTED] locate a2billing.php /usr/src/a2billing/Chameleon/A2Billing_AGI/a2billing.php /var/lib/asterisk/agi-bin/a2billing.php Could be because of the missing pcntl php extension? [EMAIL PROTECTED] rpm -qa | grep php php-mysql-4.3.9-3 php-ldap-4.3.9-3 php-odbc-4.3.9-3 php-pgsql-4.3.9-3 php-4.3.9-3 php-pear-4.3.9-3 Thanks Joao Pereira Vahan Yerkanian wrote: exten = _2,1,Answer exten = _2,2,Wait,2 exten = _2,3,DeadAGI, a2billing.php exten = _2,4,Wait,2 exten = _2,5,Hangup I tried it and the call is answered bu Asterisk and never dials the destination. :( Yes that's the correct way to launch A2B script. Are you a2billing.php is in your agi-bin directory? Also, you can see if the script runs without error by executing it from shell(you'll need php cli compiled and installed) and keep pressing enter key to see the script output. Perhaps you have your php binary in the wrong path or a missing php extension. Make sure you have pcntl php extension installed too. HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1443 (20060314) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] using a billing system
Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten = _2,1,Dial(SIP/[EMAIL PROTECTED]) Now, with Asterisk2Billing would be something like this? exten = _2,1,Answer exten = _2,2,Wait,2 exten = _2,3,DeadAGI,a2billing.php exten = _2,4,Wait,2 exten = _2,5,Hangup I tried it and the call is answered bu Asterisk and never dials the destination. :( What do I need to put in the Asterisk configuration in order to make the call and start the billing engine? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..
Hello Just 2 ideas: How cares about GSM WiFi handovers? I just want to make free VoIP calls. About the ISPs blocking VoIP: I believe they will not block VoIP because a lot of theire services are VoIP based, like the webcasts, the TV shows over the Internet, and all the multimedia stuff they want us to buy. In Portugal I already did 3G VoIP calls from TMN and Vodafone. I would really like to try this phone :) Regards Joao Pereira Steve Kennedy wrote: On Tue, May 23, 2006 at 02:50:33AM +0800, Sam Tam wrote: Well it is incorrect to say that. In places like USA or London, a lot of areas are covered by local wifi providers, some are free, some aren't. You then can use them to drop some of your local or international calls cheaply by using wifi. But the point is without operator cooperation, there's no seamless handover between GSM and WiFi, and the operators don't want to lose the revenue on the voice, so they are unlikely to support it. BT have an arrangement with Vodafone for their Fusion service (using an in-premise Bluetooth basestation and a phone with GSM/Bluetooth), but they're big enough to force an operator's hand. For general GSM/WiFi UMA, it's unlikely the (UK) operators will allow other providers access to their networks, as it reduces their revenues. They're already p*ssed off enough that they're being forced to reduce roaming charges (currently on voice - but the EU is likely to look at data charges which can be extremely costly). They are desperate to keep revenues. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VoIP provider for Asterisk
Hi I dont know if it's the best, but for Portugal and to place calls throwout Europe, www.startel.pt has a good service. Regards Joao Kerry Garrison wrote: Depends on your location and your requirements. A generic post like this generally turns into a flame war. Please be MUCH more specific. -Kerry *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Crazy Boy *Sent:* Tuesday, May 23, 2006 5:56 AM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] Best VoIP provider for Asterisk Hi Friends, Can you please tell me who is the best VoIP Service Provider using Asterisk (With trail version for sometime) . Waiting for your quick response. Thank you. Regards, Chandra. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] doing SIP URI calls
Hello to all
Im trying to make SIP URI calls with my [EMAIL PROTECTED], and I followed this:
http://slacker.com/~nugget/projects/asterisk/page7
So I putted in extensions.conf:
MYDOMAIN = xxx.xxx.xxx.xxx
MYFQDN = xxx.xxx.xxx.xxx
[macro-uridial]
exten = s,1,NoOp(Outbound SIP URI call ${ARG1})
exten = s,2,SetCIDNum(5125380508)
exten = s,3,Dial(SIP/${ARG1})
exten = s,4,Congestion()
and in extensions_custom.conf :
[from-internal-custom]
exten = _.,1,NoOp(Incoming Call from house extension ${CALLERID} for
[EMAIL PROTECTED])
exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _.,6,NoOp(@${SIPDOMAIN} is remote, forwarding...)
exten = _.,7,Macro(uridial,[EMAIL PROTECTED])
exten = _.,8,HangUp()
exten = _.,10,Goto(noturi,${EXTEN},1)
exten = h,1,HangUp()
[noturi]
include = local
include = trunkld
include = trunkint
include = emergency
Then, I try to call [EMAIL PROTECTED] and the call fails:
asterisk debug:
Looking for 613 in from-internal (domain fwd.pulver.com)
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
SIP/2.0 404 Not Found
If I have _. in [from-internal-custom] why do the call fails?
Thanks
Joao Pereira
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[Asterisk-Users] LDAPget
Hello to all Im using [EMAIL PROTECTED] 2.7 and I would like to do LDAP querys. Can I simply use LDAPget or do I need to install Asterisk::LDAP from Alkaloid Networks? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] do extensions must be numbers in [EMAIL PROTECTED]
Hello to all In Asterisk, SIP clients can be registered with numbers (2001, 2002, ...) or with names (manuel, maria,...). But [EMAIL PROTECTED] only allows SIP registers to be done with numbers... Is there any way of register SIP users with names and then give them a numeric alias? Thanks Joao ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP domain in Asterisk
Hello to all Can someone tell me if its possible to implement a SIP domain with Asterisk (im trying with [EMAIL PROTECTED]). With a SIP domain I mean: -users having URIs with [EMAIL PROTECTED] ( instead of [EMAIL PROTECTED] ) -being able to reach our users anywhere in the world with SIP URIs (and the help of SRV records) -the possibility of dialing [EMAIL PROTECTED] and route the calls through the Internet Can this be done? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDRs and billing
Hello I configured Asterisk to put CDRs in the database like it was explained in: www.voip-info.org/wiki/view/Asterisk+cdr+pgsql What I want to know is how do the billing solutions (like Asterisk2Billing) work with Asterisk. The billing system just use the information that Asterisk puts in the CDR table? Or they connect directly to Asterisk? Or is Asterisk that has, before the Dial command, to put the information on the Asterisk2Billing tables? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDRs and billing
Ok, no problem, Ill do it with the AGI. Do I need to re-compile asterisk to support the AGI writing? or it goes by default? Thank you Joao Pereira Chris Mason (Lists) wrote: Joao Pereira wrote: Hello I configured Asterisk to put CDRs in the database like it was explained in: www.voip-info.org/wiki/view/Asterisk+cdr+pgsql What I want to know is how do the billing solutions (like Asterisk2Billing) work with Asterisk. The billing system just use the information that Asterisk puts in the CDR table? Or they connect directly to Asterisk? Or is Asterisk that has, before the Dial command, to put the information on the Asterisk2Billing tables? Asterisk2Billing requires you route the calls to its AGI, and it keeps its own database, so what you did is of no use for billing. I haven't found an application that bills from the CDRs, everything I found wanted to create the database entries. I think ASTPP can read your CDR, though. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] billing with PostgreSQL
Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] the best billing tool for Asterisk
Hello to all I would like to know some opinions of people that are using billing tools for Asterisk. Can you please advise me in wich billing tool to I use? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Routing SIP calls via URI
But is there a way of doing this without a prefix?
because people should dial without prefixes: [EMAIL PROTECTED] , not like:
[EMAIL PROTECTED]
How can we make this without a prefix? something like:
if( !uri=~@mydomain.pt ){
forward the all to the Internet
}
:)
Thanks
Joao Pereira
Shad Mortazavi wrote:
Dear Group,
I was able to fix this problem;
The solution was to use a prefix to dial out.
The next challenge was to send the SIP Domain over IAX2!. I found that
if I included @SIPDOMAIN it would break the IAX2 communications.
exten = _6.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/[EMAIL PROTECTED]),
breakes because @SIPDOMAIN is treated as the target context. You also
can not include @Context after the @SIPDOMAIN.
I created a new variable DS which was a concatenation of EXTEN and
SIPDOMAIN separated by % and not @ and I was now able to pass this over
IAX2;
DS = EXTEN%SIPDOMAIN.
exten = _6.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${DS}).
At the other end I used the CUT command and substring facilities in
Asterisk to split DS by the % eliminator; I re-formed a new variable
which was
DS = [EMAIL PROTECTED]
I can now pass calls from my internal Asterisk server to my external
Asterisk server using IAX2 and then call any external VoIP number.
Warm Regards
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc
-Original Message-
From: Shad Mortazavi
Sent: Thursday, March 30, 2006 10:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten = shad,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
This answers part of the question;
However what I want to do is to send any outbound sip calls via our
external SIP server.
i.e;
VPN LANIAX2DMZ Internet
Internal UA --- Internal (*) -- External (*)--
ExternalUA
We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
for Voicemail, 2xxx for Meetme, etc.
Do I need to setup a prefix to dial the internet? And then route all
calls to the External(*) based on this prefix?
Thanks
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc
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[Asterisk-Users] Asterisk behind NAT
Hello to all Can we put Asterisk in a company that has an ADSL connection with just one public IP address? Because with just one public IP, Asterisk must have a private (NATed) IP... but the idea is to make him dial other SIP domains. Can Asterisk work behing NAT, and still route calls to the Internet? And he can still receive calls from the Internet? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT
Thank you very much And if I put the correct SRV records in the DNS, can Asterisk receive calls?? How does the router knows, that the call must be delivered to Asterisk? Can I map all the requests that reach the router port 5060, to be delivered in 192.168.0.50 ? Did someone implemented successfully a SIP domain in Asterisk behind NAT? Thanks Joao Pereira Kerry Garrison wrote: Yes. In Sip.conf you need the following lines: externip=xxx.xxx.xxx.xxx ; put public ip address here localnet=192.168.10.0/255.255.255.0 ; edit as appropriate In your firewall, add the following mappings to your server: 5060-5061 UDP 10,000 - 20,000 UDP Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Thursday, April 06, 2006 8:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk behind NAT Hello to all Can we put Asterisk in a company that has an ADSL connection with just one public IP address? Because with just one public IP, Asterisk must have a private (NATed) IP... but the idea is to make him dial other SIP domains. Can Asterisk work behing NAT, and still route calls to the Internet? And he can still receive calls from the Internet? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - force Cisco phones to reboot
I dont have this cisco-check-cfg exten command in my asterisk... Did you installed some extra module or channel? Thanks Joao Pereira Aaron Daniel wrote: It really depends on the number of phones you're wanting to reboot. Whenever we do a reconfiguration of our phones, I have a script that runs that night that pulls all the names from the db that are cisco phones, and does a sip notify cisco-check-cfg exten in asterisk, which notifies the phone to reboot in 20 seconds if nothing interesting happens (phone call comes in... browsing the interface... stuff like that). In order for this to work, you have to put a file in the tftpboot folder called syncinfo.xml containing this: SYNCINFO IMAGE VERSION=* SYNC=0/ /SYNCINFO in order for the phones to actually reboot though. That's what we do anyway :) Aaron Joao Pereira wrote: Hello to all Does someone knows how to force the Cisco IP phones (7940 and 7960) to reboot weekly or monthly? I think this would be useful because sometimes we change the configuration settings in the TFTP, but the phone just check the TFTP when he restarts... Thanks João ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT - force Cisco phones to reboot
Hello to all Does someone knows how to force the Cisco IP phones (7940 and 7960) to reboot weekly or monthly? I think this would be useful because sometimes we change the configuration settings in the TFTP, but the phone just check the TFTP when he restarts... Thanks João ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)
And about the 802.1x ? The phones can work as passthrough and force the PC to use 802.1x ? What configuration do we put in the switches? Do we put the switch as access (with 802.1x) or trunk (without 802.1x) ? Thanks Joao Pereira Greg Oliver wrote: It actually depends on the switch model. Some put the port into trunking mode automatically with the sw voi command, and some do not. Hopefully one day Cisco will finally make their own products and become uniform instead of buying several companies and glue'ing them all together to get an ethernet switch that works. At least they got the routers right :) On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote: You don't need switchport mode trunk when using switchport voice vlan.. On 3/1/06, Nicholas Kathmann [EMAIL PROTECTED] wrote: Joao Pereira wrote: Hello to all I would like to know If some of you have already configured an Cisco IP Phone (7940 or 7960) to work in a different VLAN than the PC that is connected through the phone switch? I know that this can be done with the Skinny firmware, but I dont if it works with the SIP firmware. The Cisco technical staff told me that these phones dont support 802.1x but can work as pass-through. This way I can still use the PCs with 802.1x and the phones in the same Ethernet plug. Did someone made it with the Cisco IP phones? What configuration do I need in the phones and in the switch? Thanks Joao Pereira If configuring with Cisco switches, I'm pretty sure they pull the information for which VLAN to operate in from the switch. You have to configure the switchports on the Cisco switch like so: interface fastethernet 0/1 switchport trunk native vlan your data vlan switchport mode trunk switchport voice vlan your voice vlan spanning-tree portfast trunk etc. Thanks, Nicholas Kathmann, CISSP Kathmann Consulting, LLC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x)
Ok, but the PC has an 802.1x client that configures the VLAN when he authenticates. Is this going to pass through the phone? And will the switch accept it? Thanks Joao Pereira Wojciech Tryc wrote: Your pc has to able to support tagged vlans. The switch on the phone will pass through both tagged and untagged vlans. W - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 02, 2006 11:51 AM Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x) And about the 802.1x ? The phones can work as passthrough and force the PC to use 802.1x ? What configuration do we put in the switches? Do we put the switch as access (with 802.1x) or trunk (without 802.1x) ? Thanks Joao Pereira Greg Oliver wrote: It actually depends on the switch model. Some put the port into trunking mode automatically with the sw voi command, and some do not. Hopefully one day Cisco will finally make their own products and become uniform instead of buying several companies and glue'ing them all together to get an ethernet switch that works. At least they got the routers right :) On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote: You don't need switchport mode trunk when using switchport voice vlan.. On 3/1/06, Nicholas Kathmann [EMAIL PROTECTED] wrote: Joao Pereira wrote: Hello to all I would like to know If some of you have already configured an Cisco IP Phone (7940 or 7960) to work in a different VLAN than the PC that is connected through the phone switch? I know that this can be done with the Skinny firmware, but I dont if it works with the SIP firmware. The Cisco technical staff told me that these phones dont support 802.1x but can work as pass-through. This way I can still use the PCs with 802.1x and the phones in the same Ethernet plug. Did someone made it with the Cisco IP phones? What configuration do I need in the phones and in the switch? Thanks Joao Pereira If configuring with Cisco switches, I'm pretty sure they pull the information for which VLAN to operate in from the switch. You have to configure the switchports on the Cisco switch like so: interface fastethernet 0/1 switchport trunk native vlan your data vlan switchport mode trunk switchport voice vlan your voice vlan spanning-tree portfast trunk etc. Thanks, Nicholas Kathmann, CISSP Kathmann Consulting, LLC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)
Hello to all I would like to know If some of you have already configured an Cisco IP Phone (7940 or 7960) to work in a different VLAN than the PC that is connected through the phone switch? I know that this can be done with the Skinny firmware, but I dont if it works with the SIP firmware. The Cisco technical staff told me that these phones dont support 802.1x but can work as pass-through. This way I can still use the PCs with 802.1x and the phones in the same Ethernet plug. Did someone made it with the Cisco IP phones? What configuration do I need in the phones and in the switch? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Deploying VoIP on a WAN
Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the SBC concept. The SBC (Session Border Controller) is not a new concept since we were using it anyway when we setup a (Asterisk+SER+SIP Proxy) Box to handle the on-net dialout calls. I'm now overwhelmed with the amount of SBCs that are suggested by the vendors to implement a solution. (http://www.juniper.net/solutions/literature/solutionbriefs/351085.pdf) Can anyone drop me some lines about this? I urgently need some feedback on this. Thanks! Joao Pereira www.fccn.pt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] adress book
Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PBX making ENUM lookups
Hello
I have a Siemens HiPath and I wanted to make him do ENUM lookups.
Then I connected it to an Asterisk (with ISDN) and route all calls to
Asterisk.
Then, Asterisk does the ENUM lookup, this way:
exten= _XXX,1,BackGround(nic.at/enum-doing)
exten= _XXX,2,EnumLookup(351${EXTEN:})
exten= _XXX,3,BackGround(nic.at/enum-successful)
exten= _XXX,4,Dial(${ENUM},30,r)
But how do I configure Asterisk to deliver the call back to the
Siemens PBX, if he doesnt find an ENUM match or if the contact is offline?
(I need it because its the Siemens PBX thats connected to the PSTN)
Thanks
Joao Pereira
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[Asterisk-Users] ENUM trees
Hello I know there are 4 well known ENUM trees: e164.arpa , e164.org , e164.info and enum.org Now... to which of these should I redirect my ENUM querys? I read that e164.org is a free public ENUM root that works in a donation based system and is free for the public at large to use. Shouldnt just exist one ENUM root? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hierarchical VoIP system
And about the protocol used to create this hierarchical network? Should I use SIP (routing between SERs) or should I use IAX (routing between Asterisks)? About ENUM, Isnt the managing of the ENUM tree going to be very complicated and heavy when we reach the millions of users? Joao Jan Saell wrote: Hi there! We have kind of the same setup but are using a few number of SER boxes for the on net calls - using enum for the lookup would be a great idea so that you can get the numbers to do sip calls and move over slowly. And for the central routing voip server make the routing use SIP redirects as the central server then can handle a lot of calls as its only doing the routing decisions. Best regards jan --On Wednesday, November 30, 2005 05:45:21 PM + Joao Pereira [EMAIL PROTECTED] wrote: Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hierarchical VoIP system
Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail clients
Hello to all
I have clients registered with names (joao, manuel, etc...) and clients
registered with numbers (123, 120,...).
To make the number clients receive voicemail, I have this:
exten = _X,1,Answer
exten = _X,2,Wait(1)
exten = _X,3,VoiceMail(u${EXTEN})
exten = _X,4,Playback(vm-goodbye)
exten = _X,5,Hangup
but for the name clients I need these 5 lines for each...
exten = pereira,1,Answer
exten = pereira,2,Wait(1)
exten = pereira,3,VoiceMail(u${EXTEN})
exten = pereira,4,Playback(vm-goodbye)
exten = pereira,5,Hangup
Is there any way I can solve this? making all calls that reach this
point go to the voicemail?
Thanks
Joao
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[Asterisk-Users] Voicemail configuration
Hello,
I have my SIP clients registered with names, and I want to implement the
voicemail in my Asterisk.
I have these lines to redirect the call to the voicemail:
exten = pereira,1,Answer
exten = pereira,2,Wait(1)
exten = pereira,3,VoiceMail(u${EXTEN})
exten = pereira,4,Playback(vm-goodbye)
exten = pereira,5,Hangup
But how do I force this rule to be applied to all calls? instead of
writing these 5 lines for each of my clients ?
If I used numbers, I could do _ ... but how do I write the rule for
client names?
Thanks
Joao Pereira
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Re: [Asterisk-Users] OT: SIP firmware image for Cisco 7940 or 7960
You can download a new SIP firmware and force the Cisco IP phone to use it. Some interesting links about it: http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx Joao Daryl Johnson wrote: Sorry for the off topic message, but I am ready to give up on this 7940... I don't know what firmware version is loaded, but based on the sniffer traces it appears to be SIP 5.x or better... The problem is that I don't have any firmware files for this device. Can anyone point me in the right direction? Thanks for the help, Daryl ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco phones port range
Hi Im using Cisco IP 7940 (with SIP firmware) and I want to force him to put the media stream in some specific port. To do it I put this in the Cisco configuration file: start_media_port: 8000 end_media_port: 9000 but the Cisco IP phone boots and doesnt accept these ports, and assumes the defaults (16384-32766). Even when I put these ports directly in the phone configuration, he doesnt accept them. How can I change the RTP ports in the Cisco IP phone? ( Like in Xlite we do: System Settings- Network - Listen RTP port ) Thanks Joao Pereira ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon Diva Server query
These cards are very good, the only problem is the price... I bought one Diva Server 4BRI for 1300 Euros... its a lot... The configuration of the board is a bit hard but check this link for help: http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI Joao Armin Schindler wrote: On Fri, 18 Nov 2005, Avi Miller wrote: Armin Schindler wrote: Actually the V-4BRI should be more expensive than the 4BRI. The 'V' does mean Voice, but this card has more Voice-features besides the standard 4BRI DSP features (I think it's G.723). Thanks for that. The quote was AU$400 less for the V-4BRI, though I'll double-check that. :) Any feedback on how well these cards perform with Asterisk? These cards are very good active cards (much less interrupts than passive cards) and I never had any performance problems with them. Are there other Active QuadBRI cards easily available in Australia that I should be investigating? I cannot answer this one. Armin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IP phone NAT config
Hello, I have SER in bridging mode with two IPs (private and public). To dial the world, my Cisco IP phones must contact the SER private IP, and the call is then proxyed by SER. All other SIP clients can do it, but the Cisco phones dont What should I put in the configuration file? For now I have this: # NAT/Firewall Traversal nat_enable: 1 nat_address: voip_control_port: 5060 start_media_port: 8000 end_media_port: 9000 nat_received_processing: 0 ...but I m not really using NAT, because, for the phones, the call just goes to the private SER IP, and they dont know nothing besides that. Does someone have a setup like this with Cisco phones? Thanks Joao ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Illegal redirection
Hello I have two Cisco 7940 phones with private addresses (10.0.11.239 and 10.0.11.140) connected to Asterisk. Asterisk is also with private address (10.0.0.135), but in another network. Between the networks I have a Checkpoint Firewall-1NG The Cisco IP phones can register because the REGISTER packets arent blocked. But the INVITEs never reach Asterisk , because the Firewall drops them, saying there was an illegal redirection. The most strange part, is that, when I try to make a phone call from PhoneA(10.0.11.239) to PhoneB(10.0.11.240), the INVITE is dropped before reaching Asterisk, and it says Illegal redirection 10.0.0.135-10.0.11.240. How can the firewall know that the INVITE was going to be redirected by Asterisk to PhoneB(10.0.11.240) Joao Pereira ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialplan defenition (closer)
The IP - pbx extension calls are already workin fine. Now Im just configuring the pbx extension - IP calls this way: [pbx extensions] --- [SIEMENS PBX] [ASTERISK] --- [SER] --- [sip clients] Thats why the Dial is for SIP only. Now Im going to try to get the 118 in Asterisk, because the 74 part is being eaten somewere. Joao Pereira Armin Schindler wrote: On Wed, 10 Aug 2005, Joao Pereira wrote: Ok, I m getting to the point, This route: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) Isn't working because the dialed number isnt maching _74XXX I putted Asterisk in capi debug mode and when I dial 74118 he says: gnugk*CLI capi debug CAPI Debugging Enabled -- CONNECT_IND ID=001 #0x0004 LEN=0078 Controller/PLCI/NCCI= 0x401 CIPValue= 0x10 CalledPartyNumber = 81118 CallingPartyNumber = 01 83118 ... -- I believe that someware 74118 is being transformed in 118... but the number that apears in this debug is CalledPartyNumber = 81118 Yes, your number is 'transformed' somewhere. CAPI only gets the '118' to dial. 81 is just the numbering plan. How do I get this call? I already tried: exten = _81118,1,Dial(SIP/[EMAIL PROTECTED],30,r) exten = 81118,1,Dial(SIP/[EMAIL PROTECTED],30,r) Where is your dial() for the CAPI line? Here you dial SIP only?! Armin ___ 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] dialplan defenition (goooooooal)
I got it The siemens PBX is cutting the 74XXX in XXX, and thats why it wasnt working. Now, to implement my dialplan in witch all the SIP phones are 74XXX, I must put the 74 manually, and the line is: exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED],5,r) Thank you to everyone that helped me. Cheers Joao Pereira Joao Pereira wrote: The IP - pbx extension calls are already workin fine. Now Im just configuring the pbx extension - IP calls this way: [pbx extensions] --- [SIEMENS PBX] [ASTERISK] --- [SER] --- [sip clients] Thats why the Dial is for SIP only. Now Im going to try to get the 118 in Asterisk, because the 74 part is being eaten somewere. Joao Pereira Armin Schindler wrote: On Wed, 10 Aug 2005, Joao Pereira wrote: Ok, I m getting to the point, This route: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) Isn't working because the dialed number isnt maching _74XXX I putted Asterisk in capi debug mode and when I dial 74118 he says: gnugk*CLI capi debug CAPI Debugging Enabled -- CONNECT_IND ID=001 #0x0004 LEN=0078 Controller/PLCI/NCCI= 0x401 CIPValue= 0x10 CalledPartyNumber = 81118 CallingPartyNumber = 01 83118 ... -- I believe that someware 74118 is being transformed in 118... but the number that apears in this debug is CalledPartyNumber = 81118 Yes, your number is 'transformed' somewhere. CAPI only gets the '118' to dial. 81 is just the numbering plan. How do I get this call? I already tried: exten = _81118,1,Dial(SIP/[EMAIL PROTECTED],30,r) exten = 81118,1,Dial(SIP/[EMAIL PROTECTED],30,r) Where is your dial() for the CAPI line? Here you dial SIP only?! Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialplan defenition
Ok, but thats static routing. My architecture is this: [pbx extensions] --- [SIEMENS PBX] [ASTERISK] --- [SER] --- [sip clients] I can't put in Asterisks sip.conf the hundreds of pbx extensions (and they are always changing), I must do a dinamic forward for all 74XXX calls. I think this is realy close: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) because it seems that is everything right... but It always answer: pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid extension 's' in context 'default', but no invalid handler Joao Pereira Moises Silva wrote: its kind of weird may be the problem is the default context, i have never used the default context, i always use a specific context for each extension. Lets say you have a registered sip number 21, then you can do in sip.conf [21] someparameter=blah... etc... context=sipcontext the important thing is the parameter called 'context' it has as value 'sipcontext'. When the extension 21 calls, then the dialed number (any number the extension 21 dials) will arrive to the specified context 'sipcontext'. in sipcontext you write [sipcontext] exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) that should work. let us know if you still have problems. On 7/29/05, Joao Pereira [EMAIL PROTECTED] wrote: but everytime I dont put the s, when I try to call 74XXX, Asterisk answers : pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid extension 's' in context 'default', but no invalid handler I think it must be something like that: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) ... but it always answers: pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid extension 's' in context 'default', but no invalid handler It must be a way to do it... Thanks João Moises Silva wrote: Please read this docs: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf you need to understand what the 's' extension does. If you use it, no matter what number they have dialed, it will start at the s extensión. If i understand your goal, YOU DONT NEED the 'exten = s,1,Answer' . You have: ;exten = s,1,Answer ;exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) please replace it for: exten = _74XXX,1,Answer() exten = _74XXX,2,Dial(SIP/[EMAIL PROTECTED],30,r) best regards On 7/29/05, Joao Pereira [EMAIL PROTECTED] wrote: Ok, now ill explain my dialplan problem Goal: When Asterisk receives a 74XXX number, should send it to its peer in 193.136.252.5:5060 (SERs IP), someting like: sip:[EMAIL PROTECTED] Here is my extensions.conf and sip.conf --- EXTENSIONS.CONF [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp TRUNK=CAPI [default] ; this way he works... but always dials sip:[EMAIL PROTECTED] ... not yet what I want ;exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r) ; this way, he dials sip:[EMAIL PROTECTED] ... ;exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r) ;this way it works... but I have to dial: ; 74XXX then he gives me dialtone, and then I must dial 74XXX again... ; not yet what I want... the idea is just dial 74XXX once, withou dialtones in between ;exten = s,1,Answer ;exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) ; what must I put here to dial sip:[EMAIL PROTECTED] ??? ---SIP.CONF [general] context=default port=1720 bindaddr=193.136.252.5 insecure=very realm=fccn.pt ;defenition of SER as a peer [193.136.252.5] type=peer username=193.136.252.5:5060 host=193.136.252.5 context=from-sip canreinvite=no insecure=very Thanks Joao Pereira - Moises Silva wrote: the problem is how are you getting there? i mean, what do you have in sip.conf and please post all the relevant text in extensions.conf, not just the 'exten = blah' part, we need to know context names to see if its matching the sip.conf configuration regards On 7/28/05, Joao Pereira [EMAIL PROTECTED] wrote: I had tried that also, but it didnt work. In that case, if I dial 74118 (for example) Asterisk answers this: pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/0' sent into invalid extension 's' in context 'default', but no invalid handler I think it needs the s... but how do I put the s and route the call to [EMAIL PROTECTED] Thanks Joao Christian Victor wrote: Joao Pereira schrieb: Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r) but this way all calls go to [EMAIL PROTECTED] . Then I tried: exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r) but this way, the system tries to dial sip:[EMAIL PROTECTED] and not [EMAIL PROTECTED] like I wanted... You were
Re: [Asterisk-Users] dialplan defenition
But to have a transparent integration with VoIP and legacy, I cant make
users dial twice... or having to whait for Asterisks dialtone, and dial
the number.
I whant to dial the 74XXX from a PBX extension (74118 for example) and
the IP phone rings.
Asterisk just need to forward the 74XXX calls, thats why I think the
solution is close to this:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
... but it always answers:
pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid
extension 's' in context 'default', but no invalid handler
Why is CAPI sending it to 's' if I explicitly write
Dial(SIP/[EMAIL PROTECTED],30,r) ??
João
Matt Riddell wrote:
Joao Pereira wrote:
Hello list,
Im writing my dial plan, in witch every SIP phone begins with 74 and
has more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I
wrote this line:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
What is happening is that capi is sending it to s.
You will need to either set up an IVR, asking which number to send it to.
So, you would do the following:
exten = s,1,Answer()
exten = s,2,Background(pls-entr-extn)
exten = _74XXX,1,Dial(SIP/${EXTEN})
exten = _74XXX,2,Goto(s|1)
exten = _74XXX,102,Goto(s|1)
You will obviously need to record the pls-entr-extn sound.
You can do this by making an exten like this:
exten = 678,1,Record(pls-entr-extn)
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Re: [Asterisk-Users] dialplan defenition
yes, I know, in my extensions.conf is writen correctly. Thanks Joao Bryce Chidester wrote: On Wed, 2005-08-10 at 15:51 +0100, Joao Pereira wrote: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) Just an observation that you have an invalid address there; you have 1193 instead of 193 I believe. Fix this and I see no reason for your problem to remain. ___ 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] dialplan defenition (closer)
Ok, I m getting to the point, This route: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) Isn't working because the dialed number isnt maching _74XXX I putted Asterisk in capi debug mode and when I dial 74118 he says: gnugk*CLI capi debug CAPI Debugging Enabled -- CONNECT_IND ID=001 #0x0004 LEN=0078 Controller/PLCI/NCCI= 0x401 CIPValue= 0x10 CalledPartyNumber = 81118 CallingPartyNumber = 01 83118 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = 1c 23 9f aa 06 80 01 00 82 01 00 8b 01 00 a1 15 02 0265d 02 01 00 80 0cJOAO PEREIRA Aug 10 17:25:22 NOTICE[1086933696]: chan_capi.c:1932 capi_handle_msg: CONNECT_IND ID=001 #0x0004 LEN=0078 Controller/PLCI/NCCI= 0x401 CIPValue= 0x10 CalledPartyNumber = 81118 CallingPartyNumber = 01 83118 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = 1c 23 9f aa 06 80 01 00 82 01 00 8b 01 00 a1 15 02 0265d 02 01 00 80 0cJOAO PEREIRA Aug 10 17:25:22 WARNING[1113294272]: pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/0' sent into invalid extension 's' in context 'default', but no invalid handler -- MANUFACTURER_IND ID=001 #0x0005 LEN=0034 Controller/PLCI/NCCI= 0x401 ManuID = 0x4944 Class = 0x70f000a Function= 0x4f4a8300 ManuData= O PEREIRA81 29 00 00 00 25 1c 23 9f aa 06 80 01 00 82 01 00 8b 01 00 a1 15 02 0265d 02 01 00 80 0cJOAO PEREIRA00 00 00 00 00 00 00 00 00 00 00 00 00 Aug 10 17:25:22 ERROR[1086933696]: chan_capi.c:2137 capi_handle_msg: Command.Subcommand = 0xff.0x82 -- INFO_IND ID=001 #0x0006 LEN=0019 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x70 InfoElement = 81118 -- INFO_IND ID=001 #0x0007 LEN=0018 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x18 InfoElement = a9 83 82 -- INFO_IND ID=001 #0x0008 LEN=0015 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x8005 InfoElement = default -- ALERT_CONF ID=001 #0x0004 LEN=0014 Controller/PLCI/NCCI= 0x401 Info= 0x0 -- DISCONNECT_IND ID=001 #0x000a LEN=0014 Controller/PLCI/NCCI= 0x401 Reason = 0x3490 -- I believe that someware 74118 is being transformed in 118... but the number that apears in this debug is CalledPartyNumber = 81118 How do I get this call? I already tried: exten = _81118,1,Dial(SIP/[EMAIL PROTECTED],30,r) exten = 81118,1,Dial(SIP/[EMAIL PROTECTED],30,r) but it never worked Any ideas? Thanks Joao ___ 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] dialplan defenition
Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r) but this way all calls go to [EMAIL PROTECTED] . Then I tried: exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r) but this way, the system tries to dial sip:[EMAIL PROTECTED] and not [EMAIL PROTECTED] like I wanted... can someone help me with theese? I believe the problem is solved using the correct parameters in the Dial command, but I couldnt find it until now... Thanks Joao Pereira ___ 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] dialplan defenition
I had tried that also, but it didnt work. In that case, if I dial 74118 (for example) Asterisk answers this: pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/0' sent into invalid extension 's' in context 'default', but no invalid handler I think it needs the s... but how do I put the s and route the call to [EMAIL PROTECTED] Thanks Joao Christian Victor wrote: Joao Pereira schrieb: Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r) but this way all calls go to [EMAIL PROTECTED] . Then I tried: exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r) but this way, the system tries to dial sip:[EMAIL PROTECTED] and not [EMAIL PROTECTED] like I wanted... You were on the right way my friend. Why not try exten = _74XXX,1,Dial(SIP/$(EXTEN)@193.136.252.5,30,r) Hope that helps Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as Gateway
Hello to all
I succefully installed Asterisk and an Eicon Diva Server 4 BRI (with
CAPI) to connect to a Siemens PBX, but I still cant forward calls to the
Siemens PBX (neither receive them from the PBX).
Here s the result in the asterisk console when I try to dial the 116 PBX
phone:
-- Executing Dial(SIP/193.136.2.205:5060-fd1f,
CAPI/12345678:b116|90) in new stack
-- data = 12345678:b116
-- capi request omsn = 12345678
== found capi with omsn = 12345678
== CAPI Call CAPI[contr1/12345678]/11 with B3-- Called 12345678:b116
-- CONNECT_CONF ID=001 #0x0012 LEN=0014
Controller/PLCI/NCCI= 0x301
Info= 0x0
-- CONNECT_CONF ID=001 #0x0012 LEN=0014
Controller/PLCI/NCCI= 0x301
Info= 0x0
== received CONNECT_CONF PLCI = 0x301 INFO = 0
-- DISCONNECT_IND ID=001 #0x001b LEN=0014
Controller/PLCI/NCCI= 0x301
Reason = 0x3302
== DISCONNECT_IND PLCI=0x301 REASON=0x3302
-- CAPI Hangingup
== No one is available to answer at this time
this is my CAPI.CONF
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=12345678
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
;echosquelch=1
;echocancel=yes
;echotail=64
;callgroup=1
;deflect=12345678
devices=2
-this is my EXTENSIONS.CONF
[from-sip]
exten = _XXX,1,Dial,CAPI/12345678:b${EXTEN}|90
Does someone have an ideia of what is missing?
The Siemens PBX should forward the call to its 116 extension... but
there's no way I can debug it...
Joao Pereira
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[Asterisk-Users] Diva Server 4BRI + Asterisk ------(QSIG)------ PBX
Hello * Is someone using succesfully a Diva Server 4BRI with QSIG? I m digging hard to implement it, because DivaServer supports QSIG, but Diva Server is used with CAPI, not with Zaptel drivers, because it doesnt have a HFC-Chipset (I think). And QSIG is implemented in Zaptel Drivers. not in CAPI. Does someone have a solution for this? Are any of my assumptions wrong? Did someone ever putted a Diva Server with Asterisk and QSIG? Thanks Joao Pereira ___ 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] ETSI or QSIG
Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have samples of the configuration? I had my Asterisk connecting to a Siemens PBX with ETSI and it was working fine, but peolpe said to me that QSIG could implement more features and turn the calls between the two systems transparent for the users. And I read that QSIG could take the caller name and doesnt need to have a dialtone when is doing the system crossing. But does Asterisk supports QSIG? What are people using to connect Asterisk with the PBXs? QSIG, ETSI or something else? Thanks João ___ 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: FW: [Asterisk-Users] ETSI or QSIG
But doesnt Asterisk supports QSIG already? I just whant to know how to configure it. João George Lin wrote: Joao, We have developed some QSIG stack over asterisk. It will be a paid system. would you be interested in ? Regards George -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joao Pereira Sent: Wednesday, July 06, 2005 7:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ETSI or QSIG Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have samples of the configuration? I had my Asterisk connecting to a Siemens PBX with ETSI and it was working fine, but peolpe said to me that QSIG could implement more features and turn the calls between the two systems transparent for the users. And I read that QSIG could take the caller name and doesnt need to have a dialtone when is doing the system crossing. But does Asterisk supports QSIG? What are people using to connect Asterisk with the PBXs? QSIG, ETSI or something else? Thanks João ___ 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] ETSI or QSIG
Thanks for the help. I also have a Eicon Diva Server BRI and I know it can be used with chan_capi and asterisk, but the QSIG configuration is not direct. Of course I googled before asking to the list and I didnt found any direct explanation if QSIG is supported. Voip-info.org sais that zapata.conf is for configuration of Digium cards I also searched the list for previous statments about QSIG and I read that it isnt fully supported. If you re using an Eicon Diva Server BRI, what are you using to connect? ETSI, QSIG or someting else? Thanks João Patrick wrote: On Wed, 2005-07-06 at 15:05 +0100, Joao Pereira wrote: Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have samples of the configuration? QSIG is not an option in capi.conf. It is an option in the configuration of my Eicon Diva Server BRI card which is used by chan_capi asterisk. So I guess you could use it to connect to the Siemens. I don't have a sample config. But does Asterisk supports QSIG? Yes. Obviously you could have googled for this info yourself. You may want to do that first next time you have a question... Set signalling to qsig in zapata.conf: http://lists.digium.com/pipermail/asterisk-users/2005-February/091109.html Other info: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf http://www.voip-info.org/tiki-index.php?page=Asterisk+legacy+integration Regards, Patrick ___ 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] ETSI or QSIG
you re using an Eicon Diva Server BRI, what are you using to connect? ETSI, QSIG or someting else? I use my Eicon Diva Server card to hook up my ISDN/BRI line through ETSI and capi.conf to asterisk. I had that configuration too, but isnt QSIG better? because QSIG can send the caller name and provide more services. The calls passing with QSIG will be transparent, and dont have dialtones e in the middle of the number dialing. I dont know If I should continue in the hard task of configuring QSIG or I just give it up for ETSI Does someone knows if the QSIG task is reachable and if it is worth the time? João Pereira Regards, Patrick ___ 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] routing in extensions.conf
Thanks Stefan, you rule...
now, tell me just one more thing please,
I putted in capi.conf :
msn=12345678
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=siemens
devices=2
and in extension.conf :
[siemens]
exten = 930,1,Dial(SIP/joao)
but this means that when 930 is dialed, user joao always receives the
calls,
but I have 10 SIP users , and I whant that, after 930 have been dialed,
to dial one more number to refer to each of the SIP users. How do I put
it in extensions.conf ?
Thanks
Joao
Stefan Helbing wrote:
Hello Joao,
first I suggest you set an context string in capi.conf to lead incoming calls
into a special context to give you more flexibility (in my opinion), e.g.
context=siemens
For this you need a line [siemens] in your extensions.conf.
Then (and also in the case you use the default context for everything) you need the necessary lines in extensions.conf.
If you call the number 930 from siemens to asterisk you need a line like
exten = 930,1,DoWhatEverYouWantToDo
This line currently is missing therefor the fallback of asterisk to an s extensions. If you want to catch this, too (what I would recommend), you need an additional line
exten = s,1,DoStandardThings
Of course, this is only the minimum, there are much more possibilities
(especially if you want to do more than one thing in an extension).
Bye
Stefan
sth==Originalnachricht==
sthVon: Joao Pereira [EMAIL PROTECTED]
sthDatum: 2005-04-22 18:25:17
sthAn: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
sthBetreff: [Asterisk-Users] routing in extensions.conf
sth
sthHello all,
sthIm using chan_capi to connect from a Siemens High Path to a Aterisk,
sthwhen I call from the Asterisk clients to the Siemens PBX, it works, when
sthI call from a Siemens client to a SIP(Asterisk) client, it doesnt work
sthand says this:
sth
sth == Starting CAPI[contr1/930]/1 at default,930,1 failed so falling back
sthto exten 's'
sth == Starting CAPI[contr1/930]/1 at default,s,1 still failed so falling
sthback to context 'default'
sthApr 22 16:50:27 WARNING[1149236544]: pbx.c:1877 ast_pbx_run: Channel
sth'CAPI[contr1/930]/1' sent into invalid extension 's' in context
sth'default', but no invalid handler
sth
sthI think the problem is in the extensions.conf configuration, when the
sthSiemens calls the Asterisk, it starts ringing and nothing happens, but
sthwhat do I have to put in the extensions.conf to route the calls to the
sthcorrect SIP user?
sthThanks
sthJoao
sth
sth***
sthhere s my capi.conf
sth
sth[general]
sthnationalprefix=0
sthinternationalprefix=00
sthrxgain=0.8
sthtxgain=0.8
sth
sth[interfaces]
sthmsn=12345678
sthincomingmsn=*
sthcontroller=1
sthsoftdtmf=1
sthaccountcode=
sthcontext=default
sth;echosquelch=1
sth;echocancel=yes
sthdevices=2
sth
sth
sth***
sthhere s my extensions.conf
sth
sth[general]
sthstatic=yes
sthwriteprotect=no
sth
sth[globals]
sthCONSOLE=Console/dsp ; Console interface for demo
sthTRUNK=CAPI
sth
sth[default]
sth
sth; SIP to SIP
sthexten = 100,1,Dial(SIP/joao)
sthexten = 101,1,Dial(SIP/encoder)
sth
sth;SIP to Siemens
sthexten = 118,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 136,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 139,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 116,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 908,1,Dial,CAPI/12345678:b${EXTEN}|30
sth
sth;Siemens to SIP
sth;exten = s,1,Dial(SIP/joao) this one works, but it always dial the SIP
sthuser joao
sth
sthexten = 555,1,Dial(SIP/joao) ; ok, this is the one that doesnt work,
sthhow can I route the calls?
sth
sth
sth___
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sthhttp://lists.digium.com/mailman/listinfo/asterisk-users
sthTo UNSUBSCRIBE or update options visit:
sth http://lists.digium.com/mailman/listinfo/asterisk-users
sth
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[Asterisk-Users] routing in extensions.conf
Hello all,
Im using chan_capi to connect from a Siemens High Path to a Aterisk,
when I call from the Asterisk clients to the Siemens PBX, it works, when
I call from a Siemens client to a SIP(Asterisk) client, it doesnt work
and says this:
== Starting CAPI[contr1/930]/1 at default,930,1 failed so falling back
to exten 's'
== Starting CAPI[contr1/930]/1 at default,s,1 still failed so falling
back to context 'default'
Apr 22 16:50:27 WARNING[1149236544]: pbx.c:1877 ast_pbx_run: Channel
'CAPI[contr1/930]/1' sent into invalid extension 's' in context
'default', but no invalid handler
I think the problem is in the extensions.conf configuration, when the
Siemens calls the Asterisk, it starts ringing and nothing happens, but
what do I have to put in the extensions.conf to route the calls to the
correct SIP user?
Thanks
Joao
***
here s my capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=12345678
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=default
;echosquelch=1
;echocancel=yes
devices=2
***
here s my extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=CAPI
[default]
; SIP to SIP
exten = 100,1,Dial(SIP/joao)
exten = 101,1,Dial(SIP/encoder)
;SIP to Siemens
exten = 118,1,Dial,CAPI/12345678:b${EXTEN}|30
exten = 136,1,Dial,CAPI/12345678:b${EXTEN}|30
exten = 139,1,Dial,CAPI/12345678:b${EXTEN}|30
exten = 116,1,Dial,CAPI/12345678:b${EXTEN}|30
exten = 908,1,Dial,CAPI/12345678:b${EXTEN}|30
;Siemens to SIP
;exten = s,1,Dial(SIP/joao) this one works, but it always dial the SIP
user joao
exten = 555,1,Dial(SIP/joao) ; ok, this is the one that doesnt work,
how can I route the calls?
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[Asterisk-Users] Diva Server configuration
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Re: : [Asterisk-Users] QSIG, Asterisk and Eicon DIVA
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Re: [Asterisk-Users] sip wifi phone?
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[Asterisk-Users] ISDN board for gatewaying
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[Asterisk-Users] ISDN board for gatewaying
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[Asterisk-Users] free pocketPC softphone (toshiba e750)
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Re: [Asterisk-Users] Tie web application to VOIP
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[Asterisk-Users] B2BUA
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Re: [Asterisk-Users] Signaling / Streaming
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[Asterisk-Users] softphones
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[Asterisk-Users] Re: [Serusers] softphones
Hi I tried Xten, its very good, because it can stay in the taskbar (next to the clock) and start when windows starts, and is allways ready to receive calls. Maybe it s the best way to introduce VoIP to my company workers But theres a feature that s missing (or I couldnt find), there s no way to connect this softphone with the adress book. I think this feature is very important, because everybody has allready a big adressbook with the friends emails, and we dont want to have this adressbook replicated (windows adressbook and Xlite phonebook). Thanks Joao - Original Message - From: Walter Carter [EMAIL PROTECTED] To: 'Joao Pereira' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Friday, January 07, 2005 3:17 PM Subject: RE: [Serusers] softphones Try Xten: http://www.xten.com/index.php?menu=productssmenu=xlite Regards, WSC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Friday, January 07, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [Serusers] softphones Hi, can someone tell be about some good and free softphones? Are they easy to use by non-tecnical users? Can someone share their experience about the implementation of VoIP softphones in a company? because usualy people dont want to make changes in the way they work I would like to know a way to convince peaple in my company to use them. Thanks Joao Pereira ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ 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] chan_cornet
Hi but did anyone have ever used a Siemens HiPath PBX with Asterisk? If you made it, please tell me how... I read that chan_cornet does exist... http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.html Is there any Digium Hardware solution for the Asterisk HiPath connection? Thanks Joao Pereira - Original Message - From: Luís Palma [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 04, 2005 10:30 PM Subject: Re: [Asterisk-Users] connect Asterisk with Siemens HiPath HG1500 Hi, It doesn't tell you much but it looks like that you are not alone when trying to integrate with Siemens Hicom. It seems someone has decided to make it by himself. http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom Regards Luis Palma On Tue, 4 Jan 2005 12:26:38 -, Joao Pereira [EMAIL PROTECTED] wrote: Hi I want to know the best way to connect Asterisk to a Siemens HiPath HG1500 PBX. Until now I came out with 3 solutions: 1-Asterisk being a H.323 client of the Siemens PBX (I believe it needs Siemens licences and Digium hardware) 2-Asterisk connecting to the PSTN phones with Voice Modems (good ideia!!! but its analog... doesnt have caller information...) 3-Using RDIS interfaces to connect the Siemens PBX does someone have other ideias? Thanks Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_cornet
Hi I dont knowif Steffen's chan_cornet is working. I emailed him, but with no result. Yesterday I read this article http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom It has some solutions... but not yet a direct Asterisk-HiPath connection. But doesnt Digium have Asterisk-HiPath solutions? Joao - Original Message - From: richard Coco To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, January 05, 2005 12:13 PM Subject: Re: [Asterisk-Users] chan_cornet Hi, The HG1500 is a HiPath3000 board and i don't have experience with Asterisk and HiPath3K. What we have is an Asterisk connected to a Siemens HiPath4000 over a H.323 trunk using oh323 and the HG3550 board. It works fine. But the Siemens HG3550 only supports H.323 V2.0 (so not a lot of features are available). May be Steffen's chan_cornet will change this. Are there any news about this project?Joao Pereira [EMAIL PROTECTED] wrote: Hibut did anyone have ever used a Siemens HiPath PBX with Asterisk?If you made it, please tell me how...I read that chan_cornet does exist...http://lists.digium.com/pipermail/asterisk-users/2004-October/069559.htmlIs there any Digium Hardware solution for the Asterisk HiPath connection?ThanksJoao Pereira- Original Message -From: "Luís Palma" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"<ASTERISK-USERS@LISTS.DIGIUM.COM>Sent: Tuesday, January 04, 2005 10:30 PMSubject: Re: [Asterisk-Users] connect Asterisk with Siemens HiPath HG1500 Hi, It doesn't tell you much but it looks like that you are not alone when trying to integrate with Siemens Hicom. It seems someone has decided to make it by himself. http://www.voip-info.org/tiki-index.php?page=Siemens+Hicom Regards Luis Palma On Tue, 4 Jan 2005 12:26:38 -, Joao Pereira <[EMAIL PROTECTED]>wrote: Hi I want to know the best way to connect Asterisk to a Siemens HiPathHG1500 PBX. Until now I came out with 3 solutions: 1-Asterisk being a H.323 client of the Siemens PBX (I believe it needs Siemens licences and Digium hardware) 2-Asterisk connecting to the PSTN phones with Voice Modems (goodideia!!! but its analog... doesnt have caller information...) 3-Using RDIS interfaces to connect the Siemens PBX does someone have other ideias? Thanks Joao Pereira ___ 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 listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Yahoo! Mail - 250MB free storage. Do more. Manage less. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
