[asterisk-users] Spectralink 8002
Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 -Jon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spectralink 8002
I figured that (since the firmware is current on the phone). I just can't figure why it won't connect. I did notice the phone was showing no net found and the AP MAC (or similar) right after that traffic. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:32:19 PM Subject: Re: [asterisk-users] Spectralink 8002 If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an error. Beyond that, I'm afraid I can't be much help. On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 -Jon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spectralink 8002
It's a Netgear WG102 AP with a WFS709TP wireless controller. The controller is basically a rebranded Aruba MC-800 (don't know about the APs). I've also tried on my WRT-54G at home, and it does the same thing. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:45:55 PM Subject: Re: [asterisk-users] Spectralink 8002 The only time I've seen no net found on a spectralink phone is when it's out of range of the AP. That doesn't make sense if it just successfully connected to the TFTP server. What sort of AP are you connecting to? Could it have a security feature that disallows reconnects within a certain time frame? On Fri, Jan 14, 2011 at 3:39 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: I figured that (since the firmware is current on the phone). I just can't figure why it won't connect. I did notice the phone was showing no net found and the AP MAC (or similar) right after that traffic. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:32:19 PM Subject: Re: [asterisk-users] Spectralink 8002 If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an error. Beyond that, I'm afraid I can't be much help. On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 -Jon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] Spectralink 8002
The traffic I originally posted was all (except the DHCP request/response) that the phone did since power on. That was sniffed at the output of the wireless controller (all APs tunnel back to the controller). The wireless controller shows the phone as connected, but I haven't gone much further with troubleshooting there... A call to Polycom may be in order, but I don't know what kind of support I get as an end user. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 4:29:17 PM Subject: Re: [asterisk-users] Spectralink 8002 Do you see any attempt on the wireless controller from the phone to connect to anything on the network after the TFTP exchange? Any traffic at all on the network from the phone? Have you tried to capture packets with Wireshark or something similar? On Fri, Jan 14, 2011 at 4:15 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: It's a Netgear WG102 AP with a WFS709TP wireless controller. The controller is basically a rebranded Aruba MC-800 (don't know about the APs). I've also tried on my WRT-54G at home, and it does the same thing. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:45:55 PM Subject: Re: [asterisk-users] Spectralink 8002 The only time I've seen no net found on a spectralink phone is when it's out of range of the AP. That doesn't make sense if it just successfully connected to the TFTP server. What sort of AP are you connecting to? Could it have a security feature that disallows reconnects within a certain time frame? On Fri, Jan 14, 2011 at 3:39 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: I figured that (since the firmware is current on the phone). I just can't figure why it won't connect. I did notice the phone was showing no net found and the AP MAC (or similar) right after that traffic. -Jon - Original Message - From: MrHanMan mrhan...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 14, 2011 3:32:19 PM Subject: Re: [asterisk-users] Spectralink 8002 If memory serves, those errors you see are normal. The phone downloads the slnk_cfg.cfg to see what other files it should get. It then just downloads the first block of each file to compare with what it already has. If it is the same, it breaks the connection, which the TFTP server sees as an error. Beyond that, I'm afraid I can't be much help. On Fri, Jan 14, 2011 at 2:46 PM, Jonathan C. Bailey jbai...@co.marshall.ia.us wrote: Hello, I hope this isn't too off topic, but I'm attempting to set up a Spectralink 8002 Wifi phone with our Asterisk installation, and seem to be running into a brick well (more of a wall than others that have posted their experiences). My problem is that the phone boots, associates with the wireless, grabs an IP (tried static too - same thing), contacts the TFTP server for firmware, then says No net found and starts all over again. The phone has already sucessfully connected and downloaded firmware (the latest - 130.009) without issue. Also, each time it boots, we see the following traffic from the phone (seems to only be looking at firmware and refusing to do anything else). I've checked the admin guide, and everything seems to be set up properly. Has anyone else used these phones and had similar issues? Thanks! 0.036561 PHONE_IP - SERVER_IP TFTP Read Request, File: slnk_cfg.cfg\000, Transfer type: octet\000 0.036891 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 (last) 0.045617 PHONE_IP - SERVER_IP TFTP Acknowledgement, Block: 1 0.049106 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd.bin\000, Transfer type: octet\000 0.049416 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.068726 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.072439 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11gl3.bin\000, Transfer type: octet\000 0.072729 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.095686 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.098958 PHONE_IP - SERVER_IP TFTP Read Request, File: pd11wsd3.bin\000, Transfer type: octet\000 0.099228 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.120597 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed\000 0.123618 PHONE_IP - SERVER_IP TFTP Read Request, File: pi110001.bin\000, Transfer type: octet\000 0.123892 SERVER_IP - PHONE_IP TFTP Data Packet, Block: 1 0.145259 PHONE_IP - SERVER_IP TFTP Error Code, Code: File already exists, Message: No download needed
Re: [asterisk-users] Asterisk and Alcatel digital phone's
No problem. We've had good luck with them so far. Support is also VERY responsive (had a work around in a few hours, and a firmware upgrade to fix the issue within a day or two). -Jon - Original Message - From: Sander Naudts s.nau...@intersui.be To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 20, 2010 8:17:18 AM Subject: Re: [asterisk-users] Asterisk and Alcatel digital phone's Hi Jonathan, I already looked at their product a few weeks ago, but because Alcatel wasn't on their list of compatible devices, I left it alone. Because of your email, I went looking on their site for a second time and noticed on their blog that they're experimenting with Alcatel devices. So after emailing them, there is a chance that we could use their product for our digital Alcatel phones. So fingers crossed and thanks for the info ;) Sander -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jonathan C. Bailey Verzonden: zaterdag 18 december 2010 18:19 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Asterisk and Alcatel digital phone's There is a product from Citel (the TVA) that we're currently using with Toshiba phones. I know they also support Avaya, Nortel, and Panasonic, but am not sure if they do any other brands. They more or less convert your old digital phones to SIP. They have have full compatibility information on their website... -Jon - Original Message - From: John Novack jnov...@stromberg-carlson.org To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 18, 2010 6:48:57 AM Subject: Re: [asterisk-users] Asterisk and Alcatel digital phone's Sander Naudts wrote: Asterisk and Alcatel digital phone's Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has to be a solution: Digital alacatel phone - directly connected to Asterisk. Short answer NO! What you are calling legacy digital phones are not universal, and for many years have been integrated with the host system. This is generally true for business systems from 2 lines and six stations to large systems with hundreds of phones. Is there some hardware gateway or something we can use? The only gatewaywill be your existing switch or another of the same generation. When the switch is removed, why would the phones not be? the analog phones, if they are not special, but POTS phones that could be used anywhere on a loop start line in a business or home could be reused, but you may find that you will not want to. We looked at the Grandstream GXW4024 gateway for our analog phones but I'm not sure the digital one's can connect to that one as well. No they cannot. Better plan on replacing all the Alcatel phones with IP ones. John Novack Kind regards, Sander Naudts -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Alcatel digital phone's
There is a product from Citel (the TVA) that we're currently using with Toshiba phones. I know they also support Avaya, Nortel, and Panasonic, but am not sure if they do any other brands. They more or less convert your old digital phones to SIP. They have have full compatibility information on their website... -Jon - Original Message - From: John Novack jnov...@stromberg-carlson.org To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 18, 2010 6:48:57 AM Subject: Re: [asterisk-users] Asterisk and Alcatel digital phone's Sander Naudts wrote: Asterisk and Alcatel digital phone's Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has to be a solution: Digital alacatel phone - directly connected to Asterisk. Short answer NO! What you are calling legacy digital phones are not universal, and for many years have been integrated with the host system. This is generally true for business systems from 2 lines and six stations to large systems with hundreds of phones. Is there some hardware gateway or something we can use? The only gatewaywill be your existing switch or another of the same generation. When the switch is removed, why would the phones not be? the analog phones, if they are not special, but POTS phones that could be used anywhere on a loop start line in a business or home could be reused, but you may find that you will not want to. We looked at the Grandstream GXW4024 gateway for our analog phones but I'm not sure the digital one's can connect to that one as well. No they cannot. Better plan on replacing all the Alcatel phones with IP ones. John Novack Kind regards, Sander Naudts -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail distribution - missing messages
I figured it out... Apparently bug 18358 effects 1.6.2.15 also... https://bugs.digium.com/view.php?id=18358 -Jon - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 13, 2010 2:47:44 PM Subject: Re: [asterisk-users] Voice mail distribution - missing messages -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C. Bailey Sent: Monday, December 13, 2010 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice mail distribution - missing messages And that is more or less what I'm using.. Odd.. Maybe it's time for a bug report, although I'd like to track down WHAT is causing the issue first... -Jon This might or might not help, but you can do Core set verbose 15 Core set debug 15 And try the call again. This should give you as much information as humanly (Asterisk-ly)? Possible. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice mail distribution - missing messages
Hello, I seem to be having an issue with voice mail on Asterisk 1.6.2.15 (file storage). Whenever someone leaves a message that is distributed to another box (like VoiceMail(100010011002,u)), but the VM never gets distributed to the intended recipients. Instead, I get the following in the logs: [Dec 13 11:54:50] NOTICE[15965]: app_voicemail.c:4988 copy_message: Copying message from 1...@default to 1...@default [Dec 13 11:54:50] WARNING[15965]: app_voicemail.c:4011 base_encode: Failed to open file: /var/spool/asterisk/voicemail/default/1001/INBOX/msg.WAV: No such file or directory [Dec 13 11:54:50] NOTICE[15965]: app_voicemail.c:4988 copy_message: Copying message from 1...@default to 1...@default [Dec 13 11:54:50] WARNING[15965]: app_voicemail.c:4011 base_encode: Failed to open file: /var/spool/asterisk/voicemail/default/1002/INBOX/msg.WAV: No such file or directory My voicemail.conf looks like the following: 1000 = 1234,Group Box,,,delete=1 1001 = 1234,Name 1,na...@domain.com 1002 = 1234,Name 2,na...@domain.com I did remove the delete=1 from the group box, and the messages showed up there, but never got distributed to the other boxes (still had the same error as above). Any thoughts? -Jon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail distribution - missing messages
I assume you mean passing the context with the box number? I just tried that and no dice.. -Jon - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 13, 2010 12:30:18 PM Subject: Re: [asterisk-users] Voice mail distribution - missing messages -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C. Bailey Sent: Monday, December 13, 2010 12:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voice mail distribution - missing messages Hello, I seem to be having an issue with voice mail on Asterisk 1.6.2.15 (file storage). Whenever someone leaves a message that is distributed to another box (like VoiceMail(100010011002,u)), but the VM never gets distributed to the intended recipients. Instead, I get the following in the logs: [Dec 13 11:54:50] NOTICE[15965]: app_voicemail.c:4988 copy_message: Copying message from 1...@default to 1...@default [Dec 13 11:54:50] WARNING[15965]: app_voicemail.c:4011 base_encode: Failed to open file: /var/spool/asterisk/voicemail/default/1001/INBOX/msg.WAV: No such file or directory [Dec 13 11:54:50] NOTICE[15965]: app_voicemail.c:4988 copy_message: Copying message from 1...@default to 1...@default [Dec 13 11:54:50] WARNING[15965]: app_voicemail.c:4011 base_encode: Failed to open file: /var/spool/asterisk/voicemail/default/1002/INBOX/msg.WAV: No such file or directory My voicemail.conf looks like the following: 1000 = 1234,Group Box,,,delete=1 1001 = 1234,Name 1,na...@domain.com 1002 = 1234,Name 2,na...@domain.com I did remove the delete=1 from the group box, and the messages showed up there, but never got distributed to the other boxes (still had the same error as above). Any thoughts? -Jon If I recall correctly, this problem also existed in at least some branches of 1.4. I think the solution was to make Voicemail(100010011002) be voicemail(1...@domain.com1...@domain.com1...@domain.com). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail distribution - missing messages
1001 and 1002 work individually. 1000 receives messages fine if you remove the delete=1 (although the messages still don't get copied).. This worked fine previously in the 1.4.x series (can't remember the exact revision at the moment). - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 13, 2010 1:30:36 PM Subject: Re: [asterisk-users] Voice mail distribution - missing messages -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C. Bailey Sent: Monday, December 13, 2010 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice mail distribution - missing messages I assume you mean passing the context with the box number? I just tried that and no dice.. -Jon Dumb question - is this the first voicemail for boxes 1001 and 1002 (can you sent them a voicemail individually)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail distribution - missing messages
And that is more or less what I'm using.. Odd.. Maybe it's time for a bug report, although I'd like to track down WHAT is causing the issue first... -Jon - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 13, 2010 2:35:58 PM Subject: Re: [asterisk-users] Voice mail distribution - missing messages -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C. Bailey Sent: Monday, December 13, 2010 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice mail distribution - missing messages 1001 and 1002 work individually. 1000 receives messages fine if you remove the delete=1 (although the messages still don't get copied).. This worked fine previously in the 1.4.x series (can't remember the exact revision at the moment). On 1.4.26 I have a voicemail group setup like this: [default] Include=voicemailgroups [voicemailgroups] 6001,1,noop 6001,2,Voicemail(1...@default1...@default1...@default) YMMV -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.2.14 1.6.2.15: blind transfer works but not Xfer on aastra
Just saw this after we did the same upgrade... Take a look at the bug below (we first saw it in 1.8) - it has a work around that you can use... https://bugs.digium.com/view.php?id=18185 -Jon - Original Message - From: sean darcy seandar...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, December 10, 2010 12:47:54 PM Subject: [asterisk-users] 1.6.2.14 1.6.2.15: blind transfer works but not Xfer on aastra Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #extension# works for a blind transfer. XferextensionXfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-009c answered SIP/side-sip-009b == Spawn extension (longdistance, 145, 1) exited non-zero on 'SIP/side-sip-009b' in macro 'stdexten' == Spawn extension (longdistance, 145, 1) exited non-zero on 'SIP/side-sip-009b' Here extension 169 answers and uses #170#: -- SIP/169-00ff answered SIP/nhi-riverside-sip-00fe -- SIP/169-00ff Playing 'pbx-transfer.ulaw' (language 'en') [Dec 10 13:38:03] DEBUG[29980]: features.c:1330 builtin_blindtransfer: transferer=SIP/169-00ff; transferee=SIP/side-sip-00fe; lastapp=; lastdata=; chan=SIP/169-00ff; dstchan= [Dec 10 13:38:03] DEBUG[29980]: features.c:1333 builtin_blindtransfer: TRANSFEREE; lastapp=Dial; lastdata=SIP/169,18,rtT, chan=SIP/nhi-riverside-sip-00fe; dstchan=SIP/169-00ff [Dec 10 13:38:03] DEBUG[29980]: features.c:1335 builtin_blindtransfer: transferer_real_context=longdistance; xferto=170 [Dec 10 13:38:03] DEBUG[29980]: features.c:1349 builtin_blindtransfer: ABOUT TO AST_ASYNC_GOTO, have a pbx... set HANGUP_DONT on chan=SIP/nhi-riverside-sip-00fe == Channel 'SIP/side-sip-00fe' jumping out of macro 'stdexten' -- Executing [...@longdistance:1] Macro(SIP/side-sip-00fe, stdexten,170,SIP/170) in new stack -- Executing [...@macro-stdexten:1] Dial(SIP/side-sip-00fe, SIP/170,18,rtT) in new stack == Using SIP RTP TOS bits 184 Any thoughts? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8?
No dice on finding a fix for this. I've been looking through the bug tracker and through the config files and haven't found anything... - Original Message - From: Jonathan C. Bailey jbai...@co.marshall.ia.us To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 21, 2010 9:42:44 PM Subject: Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8? I know about the Progress command, but isn't that only for *inbound* channels? It's only outbound calls that I have an issue with. My two test chases are: SIP Phone - Asterisk - PRI ...and... Channel Bank - Asterisk - PRI -Jon - Original Message - From: Paul Belanger pabelan...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 21, 2010 9:33:21 PM Subject: Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8? On 10-11-21 09:41 PM, Jonathan C. Bailey wrote: Does anyone know what changed between 1.4 and 1.8 in regards to early audio (both hearing it and interacting with it)? Read UPGRADE.txt and CHANGES *CLI core show application Progress -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8?
Hello,
We recently upgraded to Asterisk 1.8/DAHDI 2.4/WANPipe 3.5.16. This system is
connected to a PRI where the provider requires long distance codes. Normally,
you dial, see progress and hear a tone (call is still unanswered at this
point), enter your code, and it starts ringing as a normal call. Since
upgrading, we aren't getting the tone anymore and aren't able to enter any
digits at the right place. The only workaround we've seen is to use something
like below (which works).
Does anyone know what changed between 1.4 and 1.8 in regards to early audio
(both hearing it and interacting with it)?
exten =
_1NXXNXX#,1,Dial(Local/ld${EXTEN:0:1...@trunk-ld,,D(ww${EXTEN:-4}))
; Dial the access extension
exten = _1NXXNXX#,n,Hangup
exten = _ld.,1,Answer
exten = _ld.,n,Dial(${PSTN_TRUNK}/${EXTEN:2},90)
exten = _ld.,n,Hangup
-Jon
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Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8?
I know about the Progress command, but isn't that only for *inbound* channels? It's only outbound calls that I have an issue with. My two test chases are: SIP Phone - Asterisk - PRI ...and... Channel Bank - Asterisk - PRI -Jon - Original Message - From: Paul Belanger pabelan...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 21, 2010 9:33:21 PM Subject: Re: [asterisk-users] Early audio (long distance codes) not working after upgrading to 1.8? On 10-11-21 09:41 PM, Jonathan C. Bailey wrote: Does anyone know what changed between 1.4 and 1.8 in regards to early audio (both hearing it and interacting with it)? Read UPGRADE.txt and CHANGES *CLI core show application Progress -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MAC Address prefixes of Voip equipment
All the Aastra equipment I have so far all has a 00:08:5d prefix. -Jon - Original Message - From: Frank Church voi...@googlemail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 12, 2010 10:41:16 AM GMT -06:00 US/Canada Central Subject: [asterisk-users] MAC Address prefixes of Voip equipment Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E. Does the same apply to other Linksys VoIP equipment? Is there some way VoIP equipment allow themselves to be identified by requesting data from some ports? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multicast Paging
I know this may be a bit off topic...
I'm trying to play a pre-recorded message to a group of Aastra phones using
multicast paging. I can page phone to phone without issue, but sending from one
of my servers to the phones results in garbled audio. Anyone else been able to
make this work without problem? My VLC command line is below.
cvlc -v emergency-test2.wav --norm-max-level=5 --sout
#transcode{acodec=ulaw,ab=64,channels=1,samplerate=8000}:rtp{dst=239.0.1.20,port-audio=16000,proto=udp}
-Jon
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Re: [asterisk-users] Multicast Paging
I think I may have to do that.. I'm beginning to think my idea with VLC just
won't work. BTW, we're running 1.4.28 (but so far there seems to be a backport).
- Original Message -
From: Leif Madsen leif.mad...@asteriskdocs.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 31, 2010 2:42:14 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] Multicast Paging
Jonathan C. Bailey wrote:
I know this may be a bit off topic...
I'm trying to play a pre-recorded message to a group of Aastra phones using
multicast paging. I can page phone to phone without issue, but sending from
one of my servers to the phones results in garbled audio. Anyone else been
able to make this work without problem? My VLC command line is below.
cvlc -v emergency-test2.wav --norm-max-level=5 --sout
#transcode{acodec=ulaw,ab=64,channels=1,samplerate=8000}:rtp{dst=239.0.1.20,port-audio=16000,proto=udp}
-Jon
Why not use the built in multicast paging system? :)
https://issues.asterisk.org/view.php?id=11797
It appears to exist in 1.6.2.
Leif.
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[asterisk-users] Preserve userfield on CDR on attended transfer
I'm attempting to link calls together in my CDR and would like to try to do it via the userfield. Is there any way to copy the userfield between calls when doing an attended transfer? I can't seem to find anything about it searching Google. -Jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stuck Parked Calls?
I've lurked for a while, but I think this is one of my first pleas for help.
I'm having issues where a parked call using the macro below is getting stuck.
Users park the call via a blfxfer key on an Aastra phone. If the call is a
blind transfer, it tries to park the call. If it isn't a blind transfer, it
tries to unpark the call. Only 2 extensions (2759 and 2760) are doing the
parking. The other extensions only pick up calls (by dialing the 3 digit park
code. The phone shows as in use and there is a call that I see via core show
channels. I can't seem to soft hangup the stuck channel either. Only killing
Asterisk forcefully will solve the issue. We're running Asterisk 1.4.18.
Thanks for any help!
[parallelparking]
exten = _7[89]X,1,Noop(Attempting to parallel park...)
exten = _7[89]X,n,Answer
exten = _7[89]X,n,Set(PARKINGEXTEN=${EXTEN})
exten = _7[89]X,n,GotoIf($[${BLINDTRANSFER} != ]?dopark:dounpark)
exten = _7[89]X,n(dopark),Noop(Going to try to park this call)
exten = _7[89]X,n,Set(RECALLEXTEN=${BLINDTRANSFER:4:4})
exten =
_7[89]X,n,ParkAndAnnounce(PARKED|180|Local/parkedannou...@parallelparking|parkreturn,${RECALLEXTEN},1)
exten = _7[89]X,n,Hangup
exten = _7[89]X,n(dounpark),Noop(Going to try to un-park this call)
exten = _7[89]X,n,ParkedCall(${EXTEN})
exten = _7[89]X,n,Hangup
exten = parkedannounce,1,Noop
exten = parkedannounce,n,Answer
exten = parkedannounce,n,Wait(1)
exten = parkedannounce,n,Hangup
[parkreturn]
exten = _,1,Noop(Returning Parked Call)
exten = _,n,SIPAddHeader(Alert-Info: info=${AASTRA_PARKRINGBACK})
exten = _,n,Set(CALLERID(name)=FrPark:${CALLERID(name)})
exten = _,n,Dial(SIP/${EXTEN},60)
exten = _,n,Hangup
Jonathan Bailey
Marshall County, Iowa
1 E Main St, Marshalltown, IA 50158
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Re: [asterisk-users] Stuck Parked Calls?
BTW, hate to reply to myself, but here is what core show channels shows for
the stuck call:
SIP/2754-0849ce682...@parkreturn:1Up (None)
Also, below is the core show channel on the SIP channel:
-- General --
Name: SIP/2754-0849ce68
Type: SIP
UniqueID: 1235508605.71766
Caller ID: 2754
Caller ID Name: (N/A)
DNID Digits: (N/A)
State: Up (6)
Rings: 0
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x4 (ulaw)
ReadFormat: 0x4 (ulaw)
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 132
Frames in: 0
Frames out: 0
Time to Hangup: 0
Elapsed Time: 17h54m39s
Direct Bridge: none
Indirect Bridge: none
-- PBX --
Context: parkreturn
Extension: 2760
Priority: 1
Call Group: 0
Pickup Group: 0
Application: (N/A)
Data: (None)
Blocking in: (Not Blocking)
Variables:
RTPAUDIOQOS=ssrc=724684267;themssrc=2145401849;lp=0;rxjitter=0.36;rxcount=6;txjitter=0.00;txcount=6;rlp=0;rtt=0.00
RECALLEXTEN=2760
PARKINGEXTEN=792
siptransfer_referer=2...@10.10.220.2
SIPTRANSFER=yes
SIPDOMAIN=10.10.220.2
BLINDTRANSFER=SIP/2760-b2e42b60
BRIDGEPEER=SIP/2760-b2e42b60
DIALEDPEERNUMBER=2754
sipcallid=60001c104135b9967ef3d91d6649c...@10.10.220.2
SIPADDHEADER01=Alert-Info: info=Bellcore-dr4
CDR Variables:
level 1: clid=2760
level 1: src=2760
level 1: dst=792
level 1: dcontext=analog-voip
level 1: channel=SIP/2754-0849ce68
level 1: lastapp=ParkAndAnnounce
level 1:
lastdata=PARKED|180|Local/parkedannou...@parallelparking|parkreturn|2760|1
level 1: start=2009-02-24 14:49:10
level 1: answer=2009-02-24 14:49:14
level 1: end=2009-02-24 14:49:14
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1235508550.71744
Jonathan Bailey
Marshall County, Iowa
1 E Main St, Marshalltown, IA 50158
- Original Message -
From: Jonathan C. Bailey jbai...@co.marshall.ia.us
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 25, 2009 8:39:42 AM GMT -06:00 US/Canada Central
Subject: [asterisk-users] Stuck Parked Calls?
I've lurked for a while, but I think this is one of my first pleas for help.
I'm having issues where a parked call using the macro below is getting stuck.
Users park the call via a blfxfer key on an Aastra phone. If the call is a
blind transfer, it tries to park the call. If it isn't a blind transfer, it
tries to unpark the call. Only 2 extensions (2759 and 2760) are doing the
parking. The other extensions only pick up calls (by dialing the 3 digit park
code. The phone shows as in use and there is a call that I see via core show
channels. I can't seem to soft hangup the stuck channel either. Only killing
Asterisk forcefully will solve the issue. We're running Asterisk 1.4.18.
Thanks for any help!
[parallelparking]
exten = _7[89]X,1,Noop(Attempting to parallel park...)
exten = _7[89]X,n,Answer
exten = _7[89]X,n,Set(PARKINGEXTEN=${EXTEN})
exten = _7[89]X,n,GotoIf($[${BLINDTRANSFER} != ]?dopark:dounpark)
exten = _7[89]X,n(dopark),Noop(Going to try to park this call)
exten = _7[89]X,n,Set(RECALLEXTEN=${BLINDTRANSFER:4:4})
exten =
_7[89]X,n,ParkAndAnnounce(PARKED|180|Local/parkedannou...@parallelparking|parkreturn,${RECALLEXTEN},1)
exten = _7[89]X,n,Hangup
exten = _7[89]X,n(dounpark),Noop(Going to try to un-park this call)
exten = _7[89]X,n,ParkedCall(${EXTEN})
exten = _7[89]X,n,Hangup
exten = parkedannounce,1,Noop
exten = parkedannounce,n,Answer
exten = parkedannounce,n,Wait(1)
exten = parkedannounce,n,Hangup
[parkreturn]
exten = _,1,Noop(Returning Parked Call)
exten = _,n,SIPAddHeader(Alert-Info: info=${AASTRA_PARKRINGBACK})
exten = _,n,Set(CALLERID(name)=FrPark:${CALLERID(name)})
exten = _,n,Dial(SIP/${EXTEN},60)
exten = _,n,Hangup
Jonathan Bailey
Marshall County, Iowa
1 E Main St, Marshalltown, IA 50158
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Re: [asterisk-users] PoE switch recommendations?
We're using D-Link DES-3028P switches (24 10/100 + 4 gbit). They also have the DES-3052P which is a 48 port version of the switch. We're paying ~$500, I think for the 24 port version from Graybar. -Jon - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 6, 2008 12:04:44 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] PoE switch recommendations? Right, it takes some doing to find a 1Gb switching phone though we ended up going with a system based on the Cisco 7941G-GE. This model supports all of the needed features including vlan tagging and 1Gb switching. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn Sent: Monday, October 06, 2008 12:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] PoE switch recommendations? Most phones support only 100M switching though Unless you run separate cabling for VoIP and data but then you would not need the 1G uplink. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gibbons Sent: Monday, October 06, 2008 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE switch recommendations? Obviously we don't need 1Gb connections for VOIP :) Phones support pass through to the desktop and VLAN tagging. The need for 1Gb ports comes from wanting to have 1Gb at the desktop. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Monday, October 06, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE switch recommendations? On Mon, 6 Oct 2008, Ken D'Ambrosio wrote: Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't care much if it's in a single chassis or not, so long as it has Gbit uplinks. I'm curious as to why you want Gb uplinks on the switches? If we assume 100Kb/sec per phone .. (gross rounding, using 100Kb/sec per phone, rather than ~80 - make the sums easier and builds in a margin) 10 calls per Mb/sec. So for a 24-port switch, 24 phones all talking to 24 extensions off that switch, the max the uplink port is going to be pushing out is 2.4Mb/sec. For 200 extensions, say 9 x 24 port switches, with a single top-level (non PoE switch) switch with the PBX plugged in along side the 9 downlinks, that single PBX link will be carrying 2.4*9 = 22Mb/sec if all phones are in-use at the same time (and the PBX is carrying media) Now you may not want to build the network like that, but it seems that Gb is overkill just for the VoIP side of things. (And with that many extensions, I would suggest keeping all the phones on one set of switches) (Then again, it might not be possible to get big PoE switches without Gb uplinks, so it might be a moot point!) So satisfy my curiosity - why Gb uplinks? Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now:
[asterisk-users] Citel Gateways
Everyone- We're looking at using some Citel gateways to serve one of our sites (40 extensions, Toshiba phones). I've found that people seem to like the product from demos, but I was wondering how many have some of the gateways in production and if they seem to do the job for the long run. -Jon ___ -- 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] Zaptel Channel Numbering
I can't seem to find anything via Google, and haven't seen this before.. What does a channel listed like Zap/0:27-1 mean? I can't figure out what the colon signifies. I seem to see channel numbers like these just before the T1 card in my Comdial switch craps itself. -Jon ___ -- 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] OT: Polycom 3.0
Polycom is affiliated with the project in some way.. They also have an official Polycom moderated vendor forum. -Jon - Original Message - From: Andreas van dem Helge [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 29, 2008 3:21:30 AM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 How do they get away with that? On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey [EMAIL PROTECTED] wrote: Try the RPM from Trixbox. If you need something to open the file on Windows, 7zip works fine.. http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html -Jon - Original Message - From: Darrick Hartman (lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 28, 2008 5:18:26 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 Andreas van dem Helge wrote: Anyone have a download link for 3.0 SIP firmware? If you are going to say ask polycom or ask your vendor don't even waste your time posting. I've asked the Nazis and they'll probably take 1 week. Suggest you get a different vendor then. I got a response from mine within a few hours. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] OT: Polycom 3.0
Try the RPM from Trixbox. If you need something to open the file on Windows, 7zip works fine.. http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html -Jon - Original Message - From: Darrick Hartman (lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 28, 2008 5:18:26 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] OT: Polycom 3.0 Andreas van dem Helge wrote: Anyone have a download link for 3.0 SIP firmware? If you are going to say ask polycom or ask your vendor don't even waste your time posting. I've asked the Nazis and they'll probably take 1 week. Suggest you get a different vendor then. I got a response from mine within a few hours. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- 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 ___ -- 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] Switch recommendation?
We've been using D-Link DES-3028P and DES-3052P switches. They can supply full power to EACH port unlike the Linksys switches we've tried. They're also rock solid from our experience. -Jon - Original Message - From: Hilary Miller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 21, 2008 8:21:12 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Switch recommendation? On Mon, Apr 21, 2008 at 5:54 PM, Sean Dennis [EMAIL PROTECTED] wrote: The Cisco 3524 switch doesn't support 802.3af which is what your Linksys phones are going to want. Thank you for sharing Sean! When I saw them I felt a disturbance in the force, and now I know why! -- Just Hil ___ -- 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 ___ -- 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] way to inquire status of T1 link
My guess is that you don't have any spans set up, or Asterisk doesn't have zaptel support... Is chan_zap.so loaded? -Jon - Original Message - From: Jerry Geis [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, April 13, 2008 1:27:56 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] way to inquire status of T1 link When I execute the commands in my cli pri show status zap show status I get errors for both commands. I am running 1.4.19, with libpri 1.4.3, and zaptel 1.4.10. how do I get these commands? Jerry -- help shows: help ! Execute a shell command abort halt Cancel a running halt ael debug contexts Enable AEL contexts debug (does nothing) ael debug macros Enable AEL macros debug (does nothing) ael debug read Enable AEL read debug (does nothing) ael debug tokens Enable AEL tokens debug (does nothing) ael nodebug Disable AEL debug messages ael reload Reload AEL configuration agent logoff Sets an agent offline agent show Show status of agents agent show online Show all online agents agi debug Enable AGI debugging agi debug off Disable AGI debugging agi dumphtml Dumps a list of agi commands in html format agi show List AGI commands or specific help cdr status Display the CDR status console answer Answer an incoming console call console autoanswer Sets/displays autoanswer console dial Dial an extension on the console console hangup Hangup a call on the console console send text Send text to the remote device core clear profile Clear profiling info core set debug channel Enable/disable debugging on a channel core set debug Set level of debug chattiness core set debug off Turns off debug chattiness core set global Set global dialplan variable core set verbose Set level of verboseness core show applications Shows registered dialplan applications core show application Describe a specific dialplan application core show audio codecs Displays a list of audio codecs core show channels Display information on channels core show channel Display information on a specific channel core show channeltypes List available channel types core show channeltype Give more details on that channel type core show codecs Displays a list of codecs core show codec Shows a specific codec core show config mappings Display config mappings (file names to config engines) core show file formats Displays file formats core show file version List versions of files used to build Asterisk core show functions Shows registered dialplan functions core show function Describe a specific dialplan function core show globals Show global dialplan variables core show hints Show dialplan hints core show image codecs Displays a list of image codecs core show image formats Displays image formats core show license Show the license(s) for this copy of Asterisk core show profile Display profiling info core show switches Show alternative switches core show threads Show running threads core show translation Display translation matrix core show uptime Show uptime information core show version Display version info core show video codecs Displays a list of video codecs core show warranty Show the warranty (if any) for this copy of Asterisk database del Removes database key/value database deltree Removes database keytree/values database get Gets database value database put Adds/updates database value database show Shows database contents database showkey Shows database contents dialplan add extension Add new extension into context dialplan add ignorepat Add new ignore pattern dialplan add include Include context in other context dialplan reload Reload extensions and * only * extensions dialplan remove extension Remove a specified extension dialplan remove ignorepat Remove ignore pattern from context dialplan remove include Remove a specified include from context dialplan save Save dialplan dialplan show Show dialplan dnsmgr reload Reloads the DNS manager configuration dnsmgr status Display the DNS manager status dundi debug Enable DUNDi debugging dundi flush Flush DUNDi cache dundi lookup Lookup a number in DUNDi dundi no debug Disable DUNDi debugging dundi no store history Disable DUNDi historic records dundi precache Precache a number in DUNDi dundi query Query a DUNDi EID dundi show entityid Display Global Entity ID dundi show mappings Show DUNDi mappings dundi show peers Show defined DUNDi peers dundi show peer Show info on a specific DUNDi peer dundi show precache Show DUNDi precache dundi show requests Show DUNDi requests dundi show trans Show active DUNDi transactions dundi store history Enable DUNDi historic records feature show Lists configured features file convert Convert audio file group show channels Display active channels with group(s) help Display help list, or specific help on a command http show status Display HTTP server status iax2 provision Provision an IAX device iax2 prune
Re: [asterisk-users] way to inquire status of T1 link
We use Nagios for network monitoring. We've got a check_pri script that should
be fairly universal. It will return critical for any alarm. Feel free to use
the script as you see fit. YMMV - may skin cats, etc (you know the disclaimer
drill)...
#! /usr/bin/python
# Checks PRI status - returns similar to the following:
# PRI span 1/0: Provisioned, Up, Active / PRI span 2/0: Provisioned, Up, Active
import os, sys, socket
statusstring = ''
for file in os.popen('/usr/sbin/asterisk -rx pri show spans').readlines():
out = file[:-1]
if out.startswith('PRI'):
statusstring += ' / ' + out.strip()
if out.startswith('Unable to connect to remote asterisk'):
print Unable to connect to Asterisk instance
sys.exit(2)
print statusstring.strip()[2:]
if statusstring.strip()[2:].count(In Alarm) 0:
sys.exit(2)
# Nagios Return Codes
# OK = 0
# Warning = 1
# Critical = 2
# Unknown = 3
sys.exit(0)
-Jon
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 12, 2008 8:21:09 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] way to inquire status of T1 link
Jerry Geis wrote:
Is there a way to inquire of the T1 link status?
I mean having cron (as example) execute a program that asks if the T1
status is OK.YEL or RED?
then on RED I can send some alert?
What sort of adaptor?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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Re: [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge?
We used it in our installation and had some issues. We were passing fax and modem calls through via the second port as a TDM bridged call. For some reason, the timing was off even though we explicitly set the timing in the redfone.conf file. We replaced it with a Sangoma A102d and haven't been happier. FWIW, it worked just fine for voice calls. It was only problematic with data communications. -Jon - Original Message - From: arkda [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 13, 2008 2:38:34 PM GMT -06:00 US/Canada Central Subject: [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge? I've been asked to look at deploying Asterisk in a high availability environment and I've been looking so I've been searching for methods to decouple the voice PRI circuits from the Asterisk server so failover to another server could take place. I've been looking at the RedFone foneBRIDGE2 2e1 product here: http://www.mapleleaf-technologies.com/webstore/redfone_fonebridge2_2e1.php Has anyone used this device (or something similar)? What were your thoughts on it? On the surface this seems like a perfect method of building high availability Asterisk environments, but I'm a little hesitant to spend a few grand just to find out it's a pipe dream. ___ -- 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 ___ -- 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] Best Console phone?
That's surprising.. When I looked at pricing, the Snom 370 was about $50 more expensive than a 57i for us (the 57i was $205). Also, configuration wasn't too bad on the Aastra, but that may just be me. BTW, it also looks like the Snom has support for an electronic headset lifter on some GN Netcom and Plantronics wireless sets. That's a cool feature that I haven't seen on the Aastra yet. -Jon - Original Message - From: Chris Bagnall [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 27, 2008 9:12:29 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Best Console phone? I'm looking for recommendations (good/bad!) on what people use for a reception console type of phone. So-far I've used Grandstreams and Snoms, both have good and bad points.. Not after anything too fancy - programmable extension buttons would be nice, good display essential... We usually use Snom 370s + sidecar modules if required. We had a brief dabble with the Aastra 57i some time ago - the LCD sidecar is beautiful - much nicer than the paper inserts in the Snom sidecar, but the Aastra was substantially more time consuming to configure, and quite a bit more expensive. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- 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 ___ -- 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] T1 Timing Troubleshooting
I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM calls (when it comes to faxing) must be having timing issues since they intermittently fail. I found what seems to be an issue in zaptel.conf (timing source for the Comdial side was 2 - changed to 0), but I don't know if that's it. I've also turned off echo cancellation. Any other thoughts on why I may be having what seem to be timing issues? Also, is timing passed through on dynamic spans bridged calls? And is there a way to verify this? Thanks! - /etc/zaptel.conf (16 channels on each PRI): loadzone=us defaultzone=us #Sangoma A400 [slot:7 bus:1 span:1] fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 fxoks=11 fxoks=12 dynamic=eth,eth1/00:50:c2:65:d0:3c/0,24,1 dynamic=eth,eth1/00:50:c2:65:d0:3c/1,24,0 # bchan=25-47 bchan=25-40 dchan=48 # bchan=49-71 bchan=49-64 dchan=72 - /etc/asterisk/zapata.conf: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes ; Turned echo cancellation off 11-15-2007 due to possible fax issues on bridged calls. echocancel=no faxdetect=no echocancelwhenbridged=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no overlapdial=yes ;Sangoma A400 [slot:7 bus:1 span:1] context=from-zaptel group=0 signalling = fxs_ks channel = 1-8 context=from-internal group=1 signalling = fxo_ks channel = 11-12 ; First port on foneBRIDGE2 - This is the PSTN side group=2 signalling = pri_cpe context=from-pstn ;channel = 25-47 (for a full PRI) ; Channels 25-40 are for a partial PRI (16 channels) channel = 25-40 ; Second port on foneBRIDGE2 - This is the Comdial side group=3 context=from-comdial signalling = pri_net ;channel = 49-71 (for a full PRI) ; Channels 49-64 are for a partial PRI (16 channels) channel = 49-64 -Jon ___ --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
