Re: [asterisk-users] G722 and Asterisk 1.6
I'd also be more sold on it if it had half the features of the GXP2000 (which is only a little over half the price). Sure, but if only half of the features in the GXP2000 actually work, what is the point of them? I'd take a stable phone with less features over one that has lots of features that don't work correctly any day. I've opened numerous tickets with Grandstream and their answer is always we will look into it and they never reply, or that doesn't work and then no reply. ___ -- 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
Re: [asterisk-users] sip prune realtime per issue
They appear to be set how I would think they need to be set: Realtime SIP Settings: -- Realtime Peers: Yes Realtime Users: Yes Cache Friends: Yes Update: Yes Ignore Reg. Expire: No Save sys. name: No Auto Clear: 120 Again, if I do a sip show peer after pruning, I see the new values, but it appears that * is still holding it somewhere that isn't updating. Marc Smith wrote: On Tue, Jul 15, 2008 at 12:05 PM, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in sip show peer , but everything is not being updated. The phone will not register even though the DB and the phone have the correct password. The only way to get it to register is to stop * and re-start it, then it works fine. I even tried changing the callerid and pruned the peer. A sip show peer shows the correct callerid, but when you call into voicemail, it is using the old callerid. Again, if I stop * and restart, it works fine. Has anybody seen this bug and if so, know what the bug ID is? We have a bunch of patches on these boxes and can't just upgrade to any old version to see if it fixes it. I need to figure out what the bug is. I did some research, but couldn't find it. Peder Do the rt* options in sip.conf have any effect? Maybe one of those might help? --Marc ___ -- 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 !DSPAM:1,487ccb5365666785646901! ___ -- 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
[asterisk-users] sip prune realtime per issue
I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in sip show peer , but everything is not being updated. The phone will not register even though the DB and the phone have the correct password. The only way to get it to register is to stop * and re-start it, then it works fine. I even tried changing the callerid and pruned the peer. A sip show peer shows the correct callerid, but when you call into voicemail, it is using the old callerid. Again, if I stop * and restart, it works fine. Has anybody seen this bug and if so, know what the bug ID is? We have a bunch of patches on these boxes and can't just upgrade to any old version to see if it fixes it. I need to figure out what the bug is. I did some research, but couldn't find it. Peder ___ -- 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
[asterisk-users] Cisco Presence
Does anybody have the settings that you use on a Cisco 7970/79x1 to get presence? I see the * side settings, but I can't find the Cisco side settings anywhere. ___ -- 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
Re: [asterisk-users] Cisco Presence
SIP. Michiel van Baak wrote: On 14:59, Wed 25 Jun 08, Peder @ NetworkOblivion wrote: Does anybody have the settings that you use on a Cisco 7970/79x1 to get presence? I see the * side settings, but I can't find the Cisco side settings anywhere. Sip or Skinny ? ___ -- 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
Re: [asterisk-users] GXW 4108 asterisk configuration
They still have issues. If you use TCP and reboot the server, the phone will never reconnect as it tries to use a closed TCP session. I opened a ticket with them and after a week their answer is . use udp. Rob Hillis wrote: Doug wrote: There is a bug in these units that won't let you put punctuation in the extension name. A Grandstream product with a bug... what an unusual concept. cough Seriously, with all the grief I've had with GXP-2000s, BT-200s and GXV-3000s, I wouldn't touch Grandstream gear with a barge pole any more. Yes, the firmware for the GXP-2000s seems to have finally stabilised, but it's taken the better part of three /years/ for this to happen. ___ -- 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
Re: [asterisk-users] [OT] wireless headphone that can answer a call?
They still make them. We use the CS70N with HL10 (headset lifter). They are around $300 with the lifter, so they aren't cheap, but they work well. The lifter fits on a Cisco 79xx phone pretty easily, but anything else requires a little extra tape and some experimentation. Peder Steve Totaro wrote: A long time ago, I used to have a desktop support role. I had a Plantronics wireless headset that actually had a transmitter, and an arm controlled by a servo that would lift the handset out of the cradle. It was crude, but worked great. I am sure there are much better integrated solutions now though. Thanks, Steve Totaro On Mon, May 5, 2008 at 8:45 AM, Andreas van dem Helge [EMAIL PROTECTED] wrote: Some of the polycom phones support this with a specific firmware and Plantronics headset. Read the polycom SIP release notes/changelog for details On Mon, May 5, 2008 at 5:29 AM, Louis-David Mitterrand [EMAIL PROTECTED] wrote: Hello and sorry for the OT, Is it possible for a wireless headset of which the base is connected to a Polycom IP601 to remotely answer a call? In the same way as a bluetooth headset. thanks, ___ -- 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] NAT issue with Fortinet Firewall
FYI, I have probably 10 Fortinet units with multiple SIP phones behind each and all of the phones work flawlessly. As long as the Fortinet is ver 3.0 or newer, it does NAT so that you don't need to have nat=yes on *. No pinholes or static nat or anything, it just works. As a side note, I probably have 20+ Cisco PIX's with the same setup and they work flawlessly too. I've seen a lot of people saying fixup sip breaks phones, but not that I have seen. I just let the PIX do nat and it works fine. Carlos Chavez wrote: I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter localnet defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have nat=yes for all phones and canreinvite=no as well. The externip parameter is set to the outside public IP address. Still we have calls with one way audio. This is the first setup with a firewall that rewrites the IP address of the endpoint so I do not know how that is affecting the packet flow. On my other servers I can always see the public IP of the endpoint. ___ -- 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] g729 encoder/decoder
How does the g729 encoder/decoder count in regards to the total number of licenses and how does it count an encoder/decoder? I looked on the wiki and don't really see anything that explains it. In other words, how do the calls below count (assume no reinvite)? g729 phone calls into voicemail g729 phone calls g711 phone g729 phone calls other g729 phone ___ -- 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] g729 encoder/decoder
That makes sense. A call from 729 to 711 would require one encoder and one decoder, right? So if you have 10 licenses, is it 10 total encoders+decoders, or 10 calls (some may require encode, or decode, or both)? Because I had 10 licenses, but my encoders+decoders was more than 10 and calls worked fine. However I also ran out of licenses when neither number was =10. Jaswinder Singh wrote: When g729 phone calls another g729 phone and you are not recording calls or doing meetme with them then license is not required ... g729 phone calling g711 will require a license to transcode the g729 side ( no license for g711 side of call ) . In short anytime u need to convert g729 into some other codec ( transcoding ) you need 1 license . On Wed, Apr 2, 2008 at 1:59 AM, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: How does the g729 encoder/decoder count in regards to the total number of licenses and how does it count an encoder/decoder? I looked on the wiki and don't really see anything that explains it. In other words, how do the calls below count (assume no reinvite)? g729 phone calls into voicemail g729 phone calls g711 phone g729 phone calls other g729 phone ___ -- 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
[asterisk-users] Grandstream BLF and Call-limit
I am trying to get BLF working on Grandstream phones with 1.2.27. I actually have it working, but I found a very strange issue and I am wondering if anybody knows what the problem is. Here is the scenario. If I have 3 phones, A, B and C. A monitors presence of B and C. Right now, if I call from B to C, B goes solid red and C flashes red, which is correct. If I add call-limit to the sip config for those phones, which the Grandstream docs show to do, and I then call from B to C. The presence for B never changes and C just goes solid red (even during ringing). The reverse holds true if I call from C to B. B shows solid red and C doesn't change from green. Any idea? If I remove call-limit on the sip.conf entries, it all goes back to working fine. I tried 2, 9 and 99 on the call-limit and they all have the same issues. I can't imagine why call-limit causes hints to stop updating correctly. Peder ___ -- 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] Help with cisco 7960 phone
Enable NAT on the phone itself and leave it enabled in *. Jerry Geis wrote: I have a cisco 7960 phone. Worked fine in the office. I took it home. At home I have a linksys router that the phone is plugged into. The linksys router has DHCP enabled. I am getting the following error on the console from the 7960. I have tried it with nat=yes and nat=no in the sip.conf file. --- Transmitting (NAT) to 192.168.1.69:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK1415360297;received=192.168.1.69;rport=5060 From: Display Name sip:[EMAIL PROTECTED];tag=1683635072 To: Display Name sip:[EMAIL PROTECTED];tag=as4c59a734 Call-ID: [EMAIL PROTECTED] CSeq: 3091 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4a1c350c Content-Length: 0 -- The username and secret are the same as they were in the office when it worked. I figure it has to be something easy but I have not found it yet. the sip.conf entry for this phone is: [570] type=friend dtmfmode=rfc2833 username=570 secret=XXX disallow=all allow=ulaw allow=alaw host=dynamic context=local-sip callerid=Home 570 570 nat=no What might I try to get the phone working from home? Thanks, Jerry ___ -- 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] Asterisk and Cisco Unity?
Do you mean Call Manager? Unity is just their voicemail system. Yes, you can use SIP to talk between * and CM. You can also use h.323, but it is a big hassle. Tony Mountifield wrote: Has anyone here any experience in getting an Asterisk box to talk to a Cisco Unity system? I have a potential customer who would like to add a conference bridge to their existing Cisco Unity setup. The digging I have done so far suggests that it should be possible to talk SIP between them, but I'd be interested in any stories of success or failure. Cheers Tony ___ -- 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] How to get a clean, basic configuration?
autoload=yes says to load everything, so you either need to change it to no and then add load statements for every module you need, or leave it as yes and then add noload for everything you don't need. Vincent wrote: On Wed, 20 Feb 2008 21:44:30 -0500, C F [EMAIL PROTECTED] wrote: vi /etc/asterisk/modules.conf Thanks, but this file doesn't hold much that's uncommented by default: # cat /etc/asterisk/modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so load = res_musiconhold.so noload = chan_alsa.so Is this really the only file that Asterisk reads to know what to load? ___ -- 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] GXP-2020 Transfer Key
What happens when you try it? And what do you do on the phone? We have lots of GXP-2000 and 2020 and transfer is one feature that does work. Gustavo Gonzalez wrote: Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with asterisk?. Attended and blind transfer does not work wiith this IP Phone Alejandro González Grupo Gestión 4384-0660 www.grupo-gestion.com.ar [EMAIL PROTECTED] --- --- RI 9000-1069 Sistema de Gestión de Calidad Certificado por IRAM Norma ISO: 9001-2000 ___ -- 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] Cisco SIP Gateway
We use PRI, not BRI, with Cisco gateways and it works great. Rock solid. Razza wrote: Is anyone using a cisco router as an ISDN gateway with Asterisk? As you might have seen from a couple of my threads, I have been looking at Fritz! and Cologne cards, both of which require development against a specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive and causes a lag in deployment. I was thinking a better approach might be to use a seperate gateway, such as a Cisco 1751 with VIC-2BRI-NT/TE talking SIP to Asterisk, much like like an SPA3K in the analogue world. Any success stories? ___ -- 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] Grandstream GXP2000 Loses Connectivity
I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a failed to register message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to get them to reconnect (I never change the config though). Has anybody seen this? I've tried lowering the Register Expiration and that seems to make it worse. If I lower it to 1 minute or 5 minutes, I lose them every 10-15 minutes. If I put it at 10 minutes, it loses connectivity once or twice a day. I tried Grandstream support and their answer was completely useless. Has anybody seen this? Or does anybody have any ideas? Again, no NAT involved, so don't say STUN or NAT issue. Peder ___ -- 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] Grandstream GXP2000 Loses Connectivity
I hate to reply to my own message, but I have some more info from debugging. A Grandstream tries to register and uses a nonce and it is accepted by *. The next time it tries to register, it uses the same none and * says SIP/2.0 401 Unauthorized. The Grandtream says ok, here try this new nonce and then it works again. Again, the next time it registers, it tries the old one and gets slapped again and gives a new one. After some indeterminate amount of time, the Grandstream actually tries to use the same nonce 3 times in a row. Once it works, the next time it gets a SIP/2.0 401 Unauthorized and then the third time, it gets a SIP/2.0 403 Forbidden. This evidently causes the Grandstream to completely give up registration as once * sends this, the Grandstream nevers tries to register again. I've waited for 1-2 hours and it never tries again. The Forbidden response appears to kill registration until the Grandstream is rebooted. Has anybody else seen this? Or maybe know how to get around it? Peder @ NetworkOblivion wrote: I did post most of that. Point to point T1, no firewalls and no nat, cisco routers, bandwidth is monitored at 30 second intervals and never exceeds 50%, almost always 25% or less. The key is that I get messages from * like failed to register and it is from the IP of the phone, so it is like the phone is sending some messed up message. Asterisk is old, 1.0.3, but it has been stable for 2-3 years with zero issues. While it could be a asterisk version issue, I have 100-150 phones on it, mostly Cisco 7940/7960 and none of them have these issues. The only phones with issues appear to be Grandstream and they are all running 1.1.5.15. Here is a sample sip config: [7834-1] context=HASKI-LD type=friend callerid=HASKI 7834 username=7834-1 secret=47834-1 host=dynamic [EMAIL PROTECTED] canreinvite=no qualify=yes Here is one of the log messages: Feb 11 18:18:05 NOTICE[20905]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.2.165' That message is from a phone that is set to register every 5 minutes. It's been 50 minutes and it still hasn't re-registered. If I reboot the phone, it will register right away... Any ideas? Peder Andrew Joakimsen wrote: Yes but network issues are still possible. What sort of network connections are you using? What sort of routers/firewalls/other network gear? Are you certain of the reliability of the T1? Also you did not post what Asterisk version is in use. Please also post the relevant sip.conf and configuration file of the phone. On Feb 11, 2008 4:52 PM, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a failed to register message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to get them to reconnect (I never change the config though). Has anybody seen this? I've tried lowering the Register Expiration and that seems to make it worse. If I lower it to 1 minute or 5 minutes, I lose them every 10-15 minutes. If I put it at 10 minutes, it loses connectivity once or twice a day. I tried Grandstream support and their answer was completely useless. Has anybody seen this? Or does anybody have any ideas? Again, no NAT involved, so don't say STUN or NAT issue. Peder ___ -- 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] Grandstream GXP2000 Loses Connectivity
I did post most of that. Point to point T1, no firewalls and no nat, cisco routers, bandwidth is monitored at 30 second intervals and never exceeds 50%, almost always 25% or less. The key is that I get messages from * like failed to register and it is from the IP of the phone, so it is like the phone is sending some messed up message. Asterisk is old, 1.0.3, but it has been stable for 2-3 years with zero issues. While it could be a asterisk version issue, I have 100-150 phones on it, mostly Cisco 7940/7960 and none of them have these issues. The only phones with issues appear to be Grandstream and they are all running 1.1.5.15. Here is a sample sip config: [7834-1] context=HASKI-LD type=friend callerid=HASKI 7834 username=7834-1 secret=47834-1 host=dynamic [EMAIL PROTECTED] canreinvite=no qualify=yes Here is one of the log messages: Feb 11 18:18:05 NOTICE[20905]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.2.165' That message is from a phone that is set to register every 5 minutes. It's been 50 minutes and it still hasn't re-registered. If I reboot the phone, it will register right away... Any ideas? Peder Andrew Joakimsen wrote: Yes but network issues are still possible. What sort of network connections are you using? What sort of routers/firewalls/other network gear? Are you certain of the reliability of the T1? Also you did not post what Asterisk version is in use. Please also post the relevant sip.conf and configuration file of the phone. On Feb 11, 2008 4:52 PM, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a failed to register message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to get them to reconnect (I never change the config though). Has anybody seen this? I've tried lowering the Register Expiration and that seems to make it worse. If I lower it to 1 minute or 5 minutes, I lose them every 10-15 minutes. If I put it at 10 minutes, it loses connectivity once or twice a day. I tried Grandstream support and their answer was completely useless. Has anybody seen this? Or does anybody have any ideas? Again, no NAT involved, so don't say STUN or NAT issue. Peder ___ -- 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
[asterisk-users] CHANUNAVAIL
I've got a setup where we have 100 DID's. Our default dialplan has one
line that calls a macro:
exten = _22XX,1,Macro(STDEXT,${EXTEN})
The macro is pretty basic:
[macro-STDEXT]
exten = s,1,NoOp
exten = s,2,Dial(SIP/${ARG1},15,Tt)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ARG1}|u)
exten = s-NOANSWER,n,Hangup
exten = s-BUSY,1,Voicemail(${ARG1}|b)
exten = s-BUSY,n,Hangup
exten = s-CHANUNAVAIL,1,Voicemail(${ARG1}|b)
exten = s-CHANUNAVAIL,n,Hangup
exten = s-CONGESTION,1,Voicemail(${ARG1}|b)
exten = s-CONGESTION,n,Hangup
My issue is that there are probably only about 50 numbers active at one
point in time and the numbers change frequently. I used the 22XX so
that I didn't have to update the dialplan all the time. The issue is
that if someone calls an invalid number, the system hangs up on them.
It tries to ring their extension, gets a SIP No Route and then goes to
CHANUNAVAIL where it tries voicemail and hangs up.
What I want to happen is if someone calls in and hits an invalid number,
it always goes to the operator. I thought I could just use CHANUNAVAIL
to send them there, but the problem is that if a phone isn't registered
and someone calls it, it goes to CHANUNAVAIL as well. This may seem
like the same thing, but it is different. If there is an extension
built and it isn't registered, I want it to go to their voicemail. It
is only if someone calls an extension that isn't built that I want it to
go to the operator.
Any ideas on how to achieve this?
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Re: [asterisk-users] CHANUNAVAIL
What about the situation where there is no voicemail box for an
extension. Is there a way to tell the difference between the phone
isn't registered and there is no phone at that extension?
Doug Lytle wrote:
exten = s-CHANUNAVAIL,1,Voicemail(${ARG1}|b)
exten = s-CHANUNAVAIL,n,Hangup
Checks for mailbox existence, if it doesn't exist, sends it to the
incoming context where further check of time of day and then on to the
operator. If it does exist, return from the Gosub and continue processing:
exten = s-CHANUNAVAIL,1,Gosub(mailbox_exist,s,1)
exten = s-CHANUNAVAIL,n,Playback(beep)
exten = s-CHANUNAVAIL,n,Voicemail([EMAIL PROTECTED]|u)
exten = s-CHANUNAVAIL,n,Hangup()
[mailbox_exist]
exten = s,1,Set(_direct_vm=${ARG1})
exten = s,n,MailboxExists([EMAIL PROTECTED])
exten = s,n,Goto(s-${VMBOXEXISTSSTATUS},1)
exten = s-FAILED,1,Answer()
exten = s-FAILED,n,Wait(1)
exten = s-FAILED,n,Playback(vm-theperson)
exten = s-FAILED,n,SayDigits(${direct_vm})
exten = s-FAILED,n,Playback(vm-nobox)
exten = s-FAILED,n,Playback(pbx-transfer)
exten = s-FAILED,n,Goto(incoming,s,1)
exten = s-SUCCESS,1,Return()
Doug
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Re: [asterisk-users] With rtcachefriends=yes, when do realtime changes take effect?
Or you can prune the specific user entry and it will look it up again. Anthony Francis wrote: Adam Moffett wrote: I asked this question last week and never got an answer. I also didn't find the answer in the wiki. I think it would be nice if asterisk would check the database again if the user re-registers, but it doesn't seem to do that. A periodic update would be ok too, but it doesn't seem to do that either. It seems like changes never happen until a reload.if that is the case then doesn't rtcachefriends completely defeat the purpose of realtime SIP 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 New entries take effect immediately, however changes require a sip reload. ___ --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] s, CDR and NoCDR in v1.4.10.1
I am running 1.4.10.1. I have a macro that is called from default for a
certain extension (both below). I added NoCDR to s to try and stop
extra CDR records, but I am still getting them. Any idea how to stop them?
extensions.conf:
[macro-STDEXT]
exten =s,1,NoCDR()
exten =s,2,Dial(${ARG1},30,Tt)
exten =s,3,Goto(s-${DIALSTATUS},1)
exten =s-NOANSWER,1,Voicemail(${ARG2}|u)
exten =s-NOANSWER,n,Hangup
exten =s-BUSY,1,Voicemail(${ARG2}|u)
exten =s-BUSY,n,Hangup
exten =s-CONGESTION,1,Voicemail(${ARG2}|u)
exten =s-CONGESTION,n,Hangup
exten =s-CHANUNAVAIL,1,Voicemail(${ARG2}|u)
exten =s-CHANUNAVAIL,n,Hangup
[default]
exten =6080,1,Macro(STDEXT,SIP/6080,6080)
Here is an example. I am getting an 's' CDR with No Answer and then an
Answered CDR in default context:
6463,6463,s,default,SIP/6080-0861a5102007-12-04
11:49:30,,2007-12-04 11:49:39,9,0,NO
ANSWER,DOCUMENTATION,,1196790570.4260,
6463,6463,6080,default,SIP/206.190.240.9-082edc08,SIP/6080-086234e0,Dial,SIP/6080|30|Tt,
2007-12-04 11:49:30,2007-12-04 11:49:39,2007-12-04
11:49:44,14,5,ANSWERED,DOCUMENTATION,,1196790570.4259,
If I don't answer, I still get an 's' CDR with No Answer. Any ideas how
to stop that? Thanks.
Peder
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[asterisk-users] SPA-2100 into Paging System Hangs
We've got an SPA-2100 connected to * and then into a paging system on one of the FXS ports. We are having an issue where the paging system doesn't hang up the line, so it stays offhook forever and obviously makes in unusable. The paging company says that the SPA needs to hangup the line once the calling user hangs up the phone. Any idea how to make it do this? It doesn't do it by default and I don't see any settings that might help with that. If I plug an analog phone in, it works just fine. Peder ___ --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] 'a' extension
Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears that the called number is 'a' and the calling number is 123. I need to be able to tell that it was a call to x456. Thanks. Peder ___ --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] (no subject)
What is the issue with the Grandstream? We are getting tired of Cisco issues, so we have started looking at Grandstream and they seem to be pretty good. The Polycom work well, but they seem to die after about a year or so. We bought 20 of them about 2 years ago and 7 of them have died or had buttons stop working so we had to replace them. I haven't had a single Cisco do that and we have probably 100 of them. Jim Houser wrote: We agree with Drew and no longer use Grandstream. We have used a few Polycom, (best voice quality, hardest to configure). I have heard good things about Snom but never used them. We standardized on Aastra. Good build, sound quality, and feature set. Easy to configure or upgrade and good pricing. If you try Snom please share your thoughts. At present we are sticking with Aastra due to good results and user feedback. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, October 31, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) [EMAIL PROTECTED] wrote: Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to our customer. The only experience we've had is a very frustrating one trying to load the IP software on a Cisco 7970G and so we assume that if we have to go through that for all 80 phones, we'll probably commit suicide :) Thanks We have used Cisco and Aastra, can't comment on Polycom or Snom. I cannot recommend Cisco, good sound quality but that's it. Ridiculously overpriced, too few usable features, incredibly awkward to manage. Aastra have good sound quality, reasonable price, configs are plain text and not to hard to work with. We have the 9133i as our basic phone and 480i in the Call Centre for the soft buttons. Both can be fed from the same config templates. We used to use Grandstream but quality and support issues have driven us away. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --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] Voicemail Options
If you setup voicemail to allow them to hit * and then it jumps to extension 'a' in the calling context, how do you see the original number that called? If each user is going to have their own jump-to number for 'a', then I have to do a db lookup based on the called number to see where to send it. If I test it and hit * from my voicemail, I get 'a' as the EXTEN, which doesn't help me. I need 'a' to be able to see the called number so that I can do a db lookup and send the call to the appropriate extension. Peder James FitzGibbon wrote: On 10/26/07, *Peder @ NetworkOblivion* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I know that you can set it up to where a user hits 0 from their mailbox and goes to an operator, but can you set up other options as well? Could I have 0 for an operator and 1 to go to another extension? I know you can do this by building an AA, but I don't want to have to do that for every user as there are about 40 people that want this. They won't all go to the same number. Thanks. You can also exit VoiceMail() using *, which jumps you to the 'a' extension in the calling context. As for building an IVR for 40 users, you could store the destination in ASTdb or realtime keyed by original extension. Then look up where to send them when they press * based upon the mailbox that VoiceMail() was called against. -- j. ___ --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] Voicemail Options
I know that you can set it up to where a user hits 0 from their mailbox and goes to an operator, but can you set up other options as well? Could I have 0 for an operator and 1 to go to another extension? I know you can do this by building an AA, but I don't want to have to do that for every user as there are about 40 people that want this. They won't all go to the same number. Thanks. Peder ___ --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] Voicemail playback on iPhone
This is semi-related, but I have a Tmobile MDA and I couldn't play the files either. The issue was not a codec issue, it was an email encoding issue. If I sent the message to an email account and it was then downloaded to my desktop via outlook and then forwarded on to my phone, I can listen to them. If I just send it direct to the phone, I see the attachment and it opens in media player, but it won't play. I don't know if you are having codec issues or email encoding issues, but it is a place to look. Incidentally, if someone knows how to get around the download email and then forward issue that I am having, I would like to hear it. Peder Jason Lixfeld wrote: I guess what I'm asking is if there is a recipe anyone has used to allow a voicemail message (once recorded by asterisk) to playback on iPhone when said recorded voicemail is received as an email attachment. I understand you can convert using sox, so I guess that's the ingredient and some sort of * configs would be the glue - I suppose it's the glue I can't figure out. I'm not trying to figure out how to get voicemails to show up in iPhone VVM or anything like that. If the voicemail configs can't be tweaked enough to record in a format iPhone can play, how can I get something like sox convert the message to another format before * emails the voicemail off to the callee? If I understand correctly, the voicemail app takes care of the entire process from the time voicemail is recorded from the caller to the time it is sent to the callee (ie: email). If that's true, then I guess I need to understand how to tell asterisk to fork from voicemail to some script to convert the recording to something iPhone friendly before we fork back to voicemail where we left off and actually email the message to the callee. Am I making any sense? On 24-Oct-07, at 11:12 AM, Baji Panchumarti wrote: Jason, I think there is a bit of terminology confusion here, you can easily convert from format to another using sox, so if your * server is going to record and email you a voicemail file, it can surely sox the file to whatever format the iphone needs it in and then send the email. It does not appear that the iPhone is using a proprietary format so just try the default recording format and see what happens. -baji. ps : I don't have an iPhone, nor have I used * voicemail yet caveat emptor :-) -- On 10/24/07, Jason Lixfeld wrote: Sorry, it's clear my question was far too vague. To clarify, is there a recipe to make * record voicemail in a format that allows playback on iPhone's media/music player playback for voicemails that are received say, in an email message. It seems the * voicemail defaults don't work. This link seems to describe codecs that do work, however I haven't seen any indications as to whether * voicemail can be tweaked to record in any of the supported formats: http://www.kehlet.cx/ Any success out there? On 22-Oct-07, at 7:38 PM, Ron Stephan wrote: Trick question I assume? It was mind numbingly simple on my iPhone...(though none of the voice mail worked when London a few weeks ago). - tap voice mail - - tap speaker (upper right) until it turns blue (is activate) - tap the message you want to playback - use assorted controls to delete - replay etc. Now...if the question is ... how do you get asterisk voice mail to show up on an iPhone...I am all ears. Groovy concept - if anybody has a hack - I'd love to see it. Elvis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Jason Lixfeld Sent: Monday, October 22, 2007 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail playback on iPhone Anyone managed to get this to work? What's the recipe? ___ ___ --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] Cisco 7940G licensing with asterisk
Yes, you need to buy a license if you use it with ANY pbx, whether it is Callmangler or Asterisk or whatever. If you buy one used, then you need to pay to re-license it as well. The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you will need a switch that provides Cisco PoE for it to work. Erick Perez wrote: Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, ___ 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] Cisco 7940G licensing with asterisk
I'm pretty sure that any Cisco switch that has PoE supports pre-standard PoE. However there are only certain ones that do support the standard. If you are looking for the cheapest used ones, then a 3524-PWR will work. If you want new, then a 3560 ps version will work. Erick Perez wrote: Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can handle the 7940G ? The 7941G does conform to the standard but it only support SCCP (shame on cisco). On 9/27/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Yes, you need to buy a license if you use it with ANY pbx, whether it is Callmangler or Asterisk or whatever. If you buy one used, then you need to pay to re-license it as well. The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you will need a switch that provides Cisco PoE for it to work. Erick Perez wrote: Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, ___ 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 ___ 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] Asterisk realtime error
Could be a mysql permission issue. Try this from the local box: mysql -u root -p enter asterisk as the password use asterisk; select * from sip_buddies; select * from iax_buddies; If you get that far and can see the entries in iax_buddies and sip_buddies, you know it isn't a permissions issue. If you can't, then you know where to look. RENZZO SOTOMAYOR wrote: Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of how to of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 NOTICE[12000]: chan_iax2.c:5252 register_verify: Host 127.0.0.1 http://127.0.0.1/ failed MD5 authentication for '101' (9a43a82001dfa49d84e8facb765f7d e2 != 31610d29241e861816b83998501ee223) I configure extconfig.conf as: [settings] iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies res_mysql.conf as: [general] dbhost = localhost dbname = asterisk dbuser = root dbpass = asterisk dbport = 3306 dbsock = /var/lib/mysql/mysql.sock My table as: CREATE TABLE iax_buddies ( name varchar(30) primary key NOT NULL, username varchar(30), type varchar(6) NOT NULL, secret varchar(50), callerid varchar(100), context varchar(100), host varchar(31) NOT NULL default 'dynamic', disallow varchar(100), allow varchar(100) ); I'm running asterisk on Fedora 6. Plz help thanks in advance Renzzo ___ 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 ___ 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] CallWithUs Service?
There has to be some reasonable priced sip provider that would perform just as well as ATT. Does it exist? The problem is that there is no QoS control between you and the provider, so a lot of the quality issues you have are probably not related to the specific provider, but just the general Internet. Until there is QoS everywhere, nobody is going to perform as well as ATT and certainly not at what everybody thinks is reasonable (1 cent per minute). ___ 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] MOH Files Volume
Is there a way to decrease the volume on the native files version of MOH in 1.4? I've had several people complain that it is too loud. Peder ___ 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] Show Callee name on Display
We have users with Cisco 7900 phones running sip. When user A calls user B, we want user B's name to appear on user A's phone. It shows the extension they call, but not the internal name of the called user. Is this possible? We have some people that used to be on an MGCP based system and they would get the callee's name popup on their phone when they called someone. I can't figure out if it is possible or if it is just a limitation of the Cisco SIP firmware. Just to clarify with an example: 1 - Steve 2 - David David calls ext 1. Right now it says calling 1. We want it to say calling Steve 1. Peder ___ 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] T1 to SIP conversion, standalone device
You can buy a used Cisco 2600 with dual-port PRI/T1 card for VoIP for ~$1500. No worries about echo-cancelation, or IRQ issues or anything like that. It just works. And the config for inbound/outbound calls is maybe 20 lines total. Alex Balashov wrote: For a price tag that does not scale to this task at all. :-) On Fri, 7 Sep 2007, Joseph Bajin wrote: Cisco AS5300/5400/5800 series Gateways should be able to do what you want as well. On 9/7/07, Alex Balashov [EMAIL PROTECTED] wrote: There are lots of these. They belong to a class of appliance known as a media gateway. http://www.voipsupply.com/product_info.php?products_id=1038 If you REALLY want to pay that kind of money for something that serves this purpose for a single T1... well, we'd all love to have your budget! :) On Fri, 7 Sep 2007, Michelle Dupuis wrote: Over a year ago I saw a discussion about a standalone device which converted a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device is? (I'm looking for a standalone device - not a PCI card). Thanks -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ 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 -- --Joe Success is easy if you think of it like Rust: It's inevitable if you keep at it. Guys claim there are magic moments, but that's just bullshit. --Fred Franzia (The famous wine guy) ___ 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ 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 ___ 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] Distributed System
The question I always have when someone mentions distributing the load across multiple machines is how do you handle contexts for phones on different machines? I want all of my phones to dial into [companyA-phones]. I have to define it in two different places (or more depending on the number of boxes). Also, say I have a single company and I want a single auto attendant with dial by name? If users go to two different boxes, then voicemail dial by name will break because voicemail won't check both boxes for the name. Also, what about dialing a peer. Say all of my phones are 2xx. If I am 201 and I dial 202, how is my dialplan setup so that it knows that 202 is on box 2, versus box 1 where I am registered? I think having several boxes works fine if you are doing home user type stuff where you don't have lots of users within one context, but if you have offices with several people, I just see lots of potential issues. I could be wrong, but I've never been able to figure out a way around it. Brian West wrote: On Aug 28, 2007, at 10:14 AM, Seysan wrote: Hi all, I'm kind a New to Asterisk.But I'm a Network Administrator with 5 years of experiance. I want to know for an installation with 90 clients, If I don't want to have just 1 server for it, then how is it possible to distribute it among about 3 servers. Should I do it in a cluster (kernel level) or something with SER? I would recommend SER plus Asterisk. I have had great success with using Asterisk with OpenSER. Best Regards, Seysan /b ___ --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] Multiple servers using realtime
I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like that. I want to use realtime for queues,voicemail, sippeers and extensions. The only issue that I have come up with so far is that a common voicemail table would cause each box to try and send out mwi indicators since it appears each * box pulls all of the voicemail boxes from the DB every 10 seconds. Any thoughts on whether I should go with one DB, or separate per box DB's? There is one mysql box, I am not referring to mysql on each box, I am referring to whether I should use separate DB's within the one mysql box for each * box. Thanks. Peder ___ --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] Queue Agents from Dialplan
Is there any way to get the channel of the first agent called in a queue? Say I have a queue with 5 agents setup in roundrobin. I want the voicemail to go to the first person that was called. Say a call comes in and rings 1,2,3, then I want it to go to vm for 1. Say the next call rings 4,5,1, I want it to go to vm for 4. I am looking for a way to get that info into the dialplan so that I can send the calls to the appropriate voicemail. Thanks. Peder ___ --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] Realtime Queue Members
Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general] realtime_family=queue_members persistentmembers = yes autofill = yes monitor-type = MixMonitor [queue2280] music = default strategy = roundrobin timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = no joinempty = yes extconfig.conf: [settings] queue_members=mysql,ASTERISK,queue_member_table MYSQL: [EMAIL PROTECTED]:/etc/asterisk# mysql -u myuser -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 7 to server version: 5.0.24a-Debian_9ubuntu2-log Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use ASTERISK; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select * from queue_member_table; ++---+-+ | queue_name | interface | penalty | ++---+-+ | queue2280 | SIP/2224 | 1 | | queue2280 | SIP/2223 | 1 | | queue2280 | SIP/ | 2 | ++---+-+ 3 rows in set (0.00 sec) I don't see any log info for mysql, except when I manually enter the info above. I've stopped an restarted * many times. I've even tried this on two separate boxes and I get the same thing. sipeers and voicemail work, but queue members does not. Any idea? I am running 1.4.10.1. Thanks. Peder ___ --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] Realtime Queue Members
Anthony Francis wrote: There is no queue_members file, asterisk doesnt know hat you are talking about, you would have to #include queue_members from inside that queue definition. Huh? How is including a file going to make realtime access the queue_members database via mysql? ___ --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] Realtime Queue Members
Thanks, that fixed it. I just looked up the bug and then patched my 1.4.10.1 source with it and it appears to work as there are now queue members listed. http://bugs.digium.com/view.php?id=10424 I can't believe nobody else ran into this. Basically the issue was that you couldn't use realtime members without having your queue in realtime queues. Now you can have a static queue with realtime members. Very useful. Peder Julian Lyndon-Smith wrote: I think that revision 80086 in the 1.4 subversion branch would fix this. Julian. Peder @ NetworkOblivion wrote: Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general] realtime_family=queue_members persistentmembers = yes autofill = yes monitor-type = MixMonitor [queue2280] music = default strategy = roundrobin timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = no joinempty = yes extconfig.conf: [settings] queue_members=mysql,ASTERISK,queue_member_table MYSQL: [EMAIL PROTECTED]:/etc/asterisk# mysql -u myuser -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 7 to server version: 5.0.24a-Debian_9ubuntu2-log Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use ASTERISK; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select * from queue_member_table; ++---+-+ | queue_name | interface | penalty | ++---+-+ | queue2280 | SIP/2224 | 1 | | queue2280 | SIP/2223 | 1 | | queue2280 | SIP/ | 2 | ++---+-+ 3 rows in set (0.00 sec) I don't see any log info for mysql, except when I manually enter the info above. I've stopped an restarted * many times. I've even tried this on two separate boxes and I get the same thing. sipeers and voicemail work, but queue members does not. Any idea? I am running 1.4.10.1. Thanks. Peder ___ --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 __ This email for dotr.com has been scanned by MessageLabs __ ___ --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] RAW asterisk!
A. BC are pre-packaged and are useful for some things, but if you deviate too much, they aren't very helpful. As a matter of fact, if you modify a text file in AsteriskNow in one of the sections that it uses, it causes the gui to freak out and it won't parse right. Plain old asterisk is a good way to learn how it really works. Bill Andersen wrote: I'm a network admin that maintains 3 commercial Asterisk servers for my employer. I am wanting to move away from the pre-packaged commercial PBXs to a more pure asterisk setup. The systems I have utilize a nice web GUI to make changes, but it really limits what I can do beyond what they have programmed into their GUI. Would I be better off starting with: a) Plain old asterisk from asterisk.org? (tutorial suggestions?) b) AsteriskNow c) Trixbox (not Pro) d) other suggestions. Thanks Bill ___ --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] Dialplan / AGI autoanswer question
First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? CallerID info is sent between the first and second ring. ___ --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] Dialplan / AGI autoanswer question
Wait(2) is what I do. Matthew Harrell wrote: First, it seems I have to have a 2 - 3 second wait before the AGI call in order to get valid CID data. Usually 2 seconds suffices for this one setup but during that time the caller has had two rings before the local extension has even begun to ring. Is there something I am doing wrong that causes it to take so long to get the CID? CallerID info is sent between the first and second ring. Well that would explain that problem, wouldn't it? Is there a proper way to wait for the CID data to be filled in if available or is Wait(2) my best option? ___ --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] Asterisk Manager to Record Greetings
FYI to anybody who cares, here is what I did: 1. Create web page where you enter a file name and a number to call 2. Insert the file name into the *DB via Asterisk Manager 3. Through Asterisk Manager create a call file from a recording extension to the phone number entered in #1 4. The recording extension answers, plays a beep, records the call to the file name that it pulls from the *DB 5. It plays the recording back and then hangs up It works perfectly. Not quite what I planned, but it does work. Doug Lytle wrote: Peder @ NetworkOblivion wrote: That's great, now say you have 5 or 6 AA's and each one has 10 different parts that you want to record (thank you for calling... for Steve press This is what I do. I found it some place on the wiki, it lets you record many prompts. exten = 4850,1,Goto(recordings,s,1) ; ** ; Welcome to the Audio prompt recording menu ; ** exten = s,1,Playback(local/extension-recording-menu) ; ; Please record your message, when ; completed press the # key ; exten = s,2,Playback(local/please-record-msg) exten = s,3,Record(mymessage:gsm) ; ; You said ; exten = s,4,Playback(local/you-said) exten = s,5,Playback(mymessage) ; *** ; Press 1 to continue or 2 to change your message ; *** exten = s,6,Background(local/press1-or-2) exten = s,7,Set(TIMEOUT(response)=2) exten = s,8,Set(TIMEOUT(digit)=2) exten = 1,1,System(/bin/mv /var/lib/asterisk/sounds/mymessage.gsm /var/lib/asterisk/sounds/local/`date +%s`.gsm) ; ; Thank you, your recording has been saved ; exten = 1,2,Playback(local/recording-saved) ; * ; Press 3 to record another message, or 4 to hangup ; * exten = 1,3,Background(local/press3-torecord-4tohang) exten = 2,1,Goto(recordings,s,2) exten = 3,1,Goto(recordings,s,2) exten = 4,1,Playback(vm-goodbye) exten = 4,2,Hangup() exten = t,1,Playback(local/sorry-didnot-getthat) exten = t,2,Goto(recordings,s,6) exten = i,1,Playback(local/sorry-invalid-choice) exten = i,2,Goto(recordings,s,2) Doug ___ --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 Manager to Record Greetings
I am trying to use Asterisk Manager via php to record auto attendant greetings and I just can't figure out how to do it. I've got the php page working and I can click to call between two phones. However if I click to call just a single phone and then try to enable monitor, when I pick up the ringing phone, it just hangs up and doesn't record anything. I'm sure I just don't know the appropriate syntax. Has anybody done something like this? I can do the php stuff, I just need the Asterisk Manager syntax. I want to call a phone, when they pick up, it starts recording to a file and when they hang up, it closes the file. Any help would be appreciated. Peder ___ --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] Asterisk Manager to Record Greetings
That's great, now say you have 5 or 6 AA's and each one has 10 different parts that you want to record (thank you for calling... for Steve press 1 for dave press 2). Rather than having to record a long message, I want to break it into pieces so that if dave leaves, we can just record that one chunk rather than the whole thing. I would need lots of extensions pre-setup for each chunk. Not very efficient. Gordon Henderson wrote: On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote: I am trying to use Asterisk Manager via php to record auto attendant greetings and I just can't figure out how to do it. I've got the php page working and I can click to call between two phones. However if I click to call just a single phone and then try to enable monitor, when I pick up the ringing phone, it just hangs up and doesn't record anything. I'm sure I just don't know the appropriate syntax. Has anybody done something like this? I can do the php stuff, I just need the Asterisk Manager syntax. I want to call a phone, when they pick up, it starts recording to a file and when they hang up, it closes the file. Any help would be appreciated. I can't help but think you're making life hard for yourself. Why not do it by dialling a code on the telephone and having the dialplan Record() what's being spoken rather than go to the bother of writing PHP to call asterisk via the monitor interface... But I don't know the whole story of your implementation! I record prompts like this: ; Record Intro message. exten = 771,1,Answer() exten = 771,2,Wait(1) exten = 771,3,Playback(beep) exten = 771,4,Record(/var/spool/app/introMessage:wav) exten = 771,5,Playback(/var/spool/app/introMessage) exten = 771,6,Hangup() Gordon ___ --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] MoH mysteriously stopped working
I've had MOH die probably 4-5 times in the last 2+ years and the only way to get it back is to stop * and restart it. Reloading MOH or just doing a regular reload doesn't work. I have to actually do a stop now and then asterisk to get it to work again. * restarts and MOH works fine. No clue why, but I have seen it on multiple versions of *. Jay Moore wrote: Folks, I have somewhat of a serious issue here. My music on hold mysteriously stopped working. I have made no changes to my Asterisk box in the past month and up until an hour ago, MoH was working fine (as far as I know). CLI: -- Started music on hold, class 'default', on channel 'IAX2/lobby-2' -- Stopped music on hold on IAX2/lobby-2 voip*CLI moh reload voip*CLI 1 class reloaded. == Destroying musiconhold processes == Parsing '/etc/asterisk/musiconhold.conf': Found Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:422 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:504 monmp3thread: Unable to spawn mp3player musiconhold.conf: - [default] mode = quietmp3 directory = /var/lib/asterisk/mohmp3 random = yes I have had .gsm (and only .gsm) files in that directory since day one, and it's always played them just fine. The .gsm files are still in that directory, and transferring them to my computer and playing them works just fine. I have autoload set in modules.conf, and I can't figure out why my music on hold suddenly stopped working. Any thoughts? Thanks in advance, Jay ___ --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] SetCallerPres and Cisco PRI
Does anybody know if SetCallerPres works on calls via SIP through a Cisco gateway? We are trying to mark outbound calls as anonymous and we set it to prohib, but calls still show outbound callerid. We are SIP from * to the Cisco gateway and then PRI outbound. If we strip the callerid num, then the first number on the PRI gets added as teh callerid, so we can't do that. We need to make it anonymous so that it shows as unknown on the other end. Peder ___ --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] Macro Goofiness
I am trying to use a macro to screen calls by calling several different
phones at the same time in different groups. Find me will not work and
queues will not work either. Trust me, I've tried them both and they
don't work like they should. Here is what I have:
A call comes into 6084 and does the following (in default context):
exten = 6084,1,NoOp(test)
exten = 6084,2,SetMusicOnHold(default)
exten = 6084,3,Dial(LOCAL/office,40,m)
exten = 6084,4,Voicemail(6084|u)
exten = 6084,5,Hangup
That calls the following, also in default context:
exten = office,1,Dial(SIP/6080,30,M(screen))
exten = office,2,Hangup
That calls the screening macro:
[macro-screen]
exten = s,1,Wait(1)
exten = s,2,Background(testmessage)
exten = s,3,WaitExten(5)
exten = s,4,NoOp(${MACRO_RESULT})
exten = h,1,Set(MACRO_RESULT=CONTINUE)
exten = h,2,NoOp(${MACRO_RESULT})
exten = i,1,Set(MACRO_RESULT=CONTINUE)
exten = i,2,NoOp(${MACRO_RESULT})
exten = t,1,Set(MACRO_RESULT=CONTINUE)
exten = t,2,NoOp(${MACRO_RESULT})
exten = 1,1,NoOp(Caller accepted)
exten = 2,1,NoOp(Pressed 2)
A call comes in, plays music on hold and calls the 6080 phone. It plays
the message and waits for a key. According to everything I've read, if
I do anything but hit 1, it should fall through and NOT bridge the
call, which should make the call go to voicemail. What happens is that
no matter what I do, it does bridge the call. If I hit 1, 2, 7 or
just wait, the call is bridged together. I've seen several samples that
are the same as this and say it should work. I can't figure it out.
I've tried it on 1.4.7 and 1.4.5 and both have the same issue.
Any ideas? If I hit a key during the WaitExten, I do NOT see the NoOp
in line 4. It's like anything just pauses execution and bridges the
call. Thanks for any tips (except people that tell me to use find
me/follow me, that won't work).
Peder
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[asterisk-users] Call Screening Not Working
I am using the Find-me/Follow-me example below with screening: http://www.voip-info.org/wiki/view/Asterisk+tips+findme Here is my actual config: [macro-screen] exten = s,1,Wait(1) exten = s,n,Background(press-1-to-be-connected-to-the-caller) exten = s,n,Set(TIMEOUT(response=5)) exten = 1,1,NoOp(Caller accepted) exten = i,1,Set(MACRO_RESULT=CONTINUE) exten = t,1,Set(MACRO_RESULT=CONTINUE) [default] exten = office,1,Dial(SIP/609,30,M(screen)) exten = office,2,Hangup exten = mobile,1,Dial(SIP/608,30,M(screen)) exten = mobile,2,Hangup exten = 6084,1,NoOp exten = 6084,2,SetMusicOnHold(default) exten = 6084,3,Dial(LOCAL/officeLOCAL/mobile,40,m) I am running 1.4.5. When I call the number, it rings the phones and plays the message, but no matter what I do, the call gets bridged. If I hit 2, or nothing, or it times out, the call gets bridged to whoever picks it up. The script should continue with the other called numbers until the timeout, but it doesn't seem to work that way. Any ideas what is wrong? My guess is that something changed in 1.4 to make this fail, but I don't really know what. ___ --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] SIP INFO message
I've got a very strange problem and I can't figure it out. I have a Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I see callerID name, but it is not getting to * via SIP. I am running * 1.4.2 and the latest Cisco IOS for my router. Here is what is happening: A call comes into the gateway. It sends a SIP INVITE to * with pending as the callerID name (this does NOT show up on any phones). * sends a TRYING message back to the gateway. * waits 2 seconds (I have a 2 second wait in the dialplan) and then sends an INVITE to the phone. The phone sends back TRYING and RINGING to *. * then sends RINGING to the gateway and the gateways sends a SIP INFO with the correct CALLERID NAME. It doesn't matter if the wait in the dialplan is 1 second, 2 seconds, or 5 seconds, it never sends the correct name until after * sends it a RINGING message. I never see any name on my display (neither pending, nor the real name). I am grabbing a tcpdump and I see pending and the real name in there, I just never see it on the * console, or on the phone. The config on * for the gateway is pretty vanilla: [192.168.1.100] context=default type=friend host=192.168.1.100 dtmfmode=rfc2833 disallow=all allow=ulaw allow=g729 canreinvite=yes qualify=yes t38pt_udptl = yes * doesn't appear to understand the INFO message as it is spitting out some errors like below, and I am dropping calls after ~ 30 seconds. [Apr 9 22:52:22] WARNING[14660]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. If I disable the feature on the gateway that sends the name, everything works fine, but I obviously don't get name. I've spoken to several other people that have virtually the same gateway config as me and theirs works. I've tried this with * 1.4.2 and 1.0.3 and I get the same results on both of them. I am to the point where I think I have some * config wrong, but I can't imagine what it could be. Anybody have any insight into why * would freak out on an INFO message? I can send Ethereal captures if that would help. Thanks. Peder ___ --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] Privacy Manager w/ No Recording
Is there a way to use privacy manager without requiring the user to enter their name? Essentially I am just looking for a way to force the called user to enter 1 to accept the call. I don't need a name recording. I want a call to come in, a message to be played, music on hold, call out to the called party, then enter 1 to accept, 2 to reject. Peder ___ --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] Privacy Manager w/ No Recording
I just opened 0009509 and used Explicit Call Acceptance as the name. Ben Klang wrote: On Monday 09 April 2007 02:48:32 pm Steve Murphy wrote: On Mon, 2007-04-09 at 09:47 -0500, Peder @ NetworkOblivion wrote: Is there a way to use privacy manager without requiring the user to enter their name? Essentially I am just looking for a way to force the called user to enter 1 to accept the call. I don't need a name recording. I want a call to come in, a message to be played, music on hold, call out to the called party, then enter 1 to accept, 2 to reject. An interesting concept. File an enhancement request on bugs.digium.com, and assign it to me, if you can; I'll look into it. I wrote a patch for Asterisk 1.0 and 1.2 which implements this as an option to app_dial. I called it Explicit Call Acceptance. I believe others have done similar things both at the application level and at the dialplan level. If you're interested, I'd be willing to resubmit my patch for consideration. /BAK/ -- Network stuff you didn't know http://www.networkoblivion.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] Cisco GW, PRI CallerID Name
Does anybody have callerid name coming in on a Cisco PRI via a Cisco gateway via SIP to *? I've seen a few people ask and a few people that say it should work, but I've never seen an actual working config. I do a debug on our Cisco gateway and I can see the callerid name, however none of the features that should send it via SIP seem to work. Cisco docs say to use the following: voice service voip signaling forward unconditional interface serial 1/0:23 isdn supp-service name calling When I enable either of those features, my calls hangup after about 30 seconds. * gives me a message [Apr 9 22:52:22] WARNING[14660]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Turning them off makes the calls work fine. signaling forward unconditional appears to be the key feature, but * doesn't seem to know what to do with the info that it is sending. There must be some way to set * to decode it, but I can't figure it out. I am running * 1.4.2 (and I've tried it on 1.0.3 and 1.2.10) and 12.4.13a on my Cisco gateway. Any ideas? FYI, my gateway has been running fine with multiple * boxes for 2+ years. I've finally decided to try and get this working, so I upgraded to 12.4.13a to see if it worked there and it still doesn't. Peder ___ --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] Multi-Level Queue
I am trying to setup a queue in a very specific way and I can't quite figure it out. I'm sure someone else has done this. I want calls to come into a queue and do a ringall on a number of phones (let's say 3). So ring them for 20 seconds or so. If there is no answer, I want it to ring a second set of phones for 20 seconds. If no answer, then go back to the first set of phones. I've seen where you could do two queues and do this, but I don't want to have to setup a second queue. I would like it all in one queue. The second part is that I want queue members to have to hit a key to accept a call. The third part is that I don't want agents to have to login. The reasoning behind all of this is that I want to ring desk phones and then if they don't answer, I want to ring cell phones. If I ring the cell phones too long, someone's voicemail will pick up, which I don't want. So if I set it up where they have to ack it, I can ring the cell phones and if someone's vm picks up, it is no big deal. Any ideas? ___ --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] Realtime call-limit
Does anybody know the sql type for the call-limit field under sip peers? Everything on voip-info is missing that entry. ___ --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] Multi-Level Queue
Kevin P. Fleming wrote: Peder @ NetworkOblivion wrote: I want calls to come into a queue and do a ringall on a number of phones (let's say 3). So ring them for 20 seconds or so. If there is no answer, I want it to ring a second set of phones for 20 seconds. If no answer, then go back to the first set of phones. I've seen where you could do two queues and do this, but I don't want to have to setup a second queue. I would like it all in one queue. This doesn't sound like a queue at all, but rather just Dial()-ing the desired extensions for that period of time. Are you really to have multiple callers (like a queue would be) or just have incoming calls ring all these phones in this pattern? This can be done with a single queue, but it will take some fancy configuration to make it work. There are a couple of reasons for what I want. 1. I want callee's to have to ack to receive the call, in case someone's cell vm picks up. 2. Yes, there could potentially be 2-4 people calling at any given point in time, so I want a sort of overflow to mobile's. 3. I don't want 5 minutes of ringing, I prefer where they get queue updates like you are the 2nd person in the queue and they hear music, rather than ringing. I guess I could have two queue's and just have it bounce back and forth between office phones and cell phones, but won't they get updates like you are th first person and then they switch to the other queue and you are the second person I also had a question about acking a call. It appears that acking a call is under agents.conf. I want to specify members as SIP/1234, etc, rather than having users login all the time. I don't want to have to login from my cell, I would prefer it to just know that my cell number is always a member. Is it possible to force an ack of a call if I define members as SIP/? ___ --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] Need feedback on vitelity
I did a trial as a wholesale provider and it seemed to work pretty good, but I could never get them to activate our account. I emailed the sales guy probably five times over a month to go ahead and fire it up and he never responded. Also, their tech support is horrible So basically they are like every other pay per minute VoIP provider Mail list wrote: Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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
Re: [asterisk-users] Pickup some else's call
Group pickup / call pickup is the feature you want.You put everybody in a group and if you want to grab a ringing phone, you just hit the group pickup code. http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups Rob Schall wrote: Question: Is it possible to pickup someone else's call who didn't park a call? My boss would like to see a way to pick up some one else's incoming call if they aren't at their desk and it's not forwarded. So if my phone were ringing and he knew i ran down the hall, he could press some key combo and give my extension, and it would transfer that incoming call to him. Possible? Rob ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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
Re: [asterisk-users] DST changes for the US
SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ I'm pretty sure this is wrong: tcpIpApp.sntp.daylightSavings.start.date=8 Should be: tcpIpApp.sntp.daylightSavings.start.date=2 which indicates second week of month. ___ --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] Realtime Extensions and Include
Is it possible to use the include command to include other contexts if you are using realtime for extensions? I've searched voip-info and some people were asking about it, but I didn't find a real answer anywhere. ___ --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] Realtime Extensions and Include
Not really what I mean. I have customer contexts, say customer1 and customer2. I also have a LD, Local and Intl context. To allow customer1 to dial LD, I include the LD context within the customer1 context. I want to skip text files and move to realtime for extensions and I want to know if I can include other contexts in the realtime mechanism like I do with the text files. Rob Schall wrote: Not sure if this is what you mean But we have includes in our sip,extensions and voicemail files. ;#include sip.inc We keep them commented out only because they are a copy of what is running in realtime. Every night the include files are generated and put in /etc/asterisk. If MySQL were to ever fail, we could just uncomment those 3 includes and reload asterisk. It wouldn't have the same dynamic nature to it, but it would bring functionality back online. Rob Peder @ NetworkOblivion wrote: Is it possible to use the include command to include other contexts if you are using realtime for extensions? I've searched voip-info and some people were asking about it, but I didn't find a real answer anywhere. ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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
Re: [asterisk-users] OT - IP Network Call Recording
Check out CallRex, they list Talkswitch as a supported product (also Asterisk): http://www.telrex.com/callrex.htm I've seen it being used with Cisco phones on a hosted Covad environment and it is pretty neat. (I have no affiliation with them whatsoever). Cory Andrews wrote: Apologies in advance as this is not directly Asterisk related, however I thought I might be able to leverage the experience of particiapants on this listserv for some advice. I have a client who is utilizing Talkswith PBX appliances, which have no native call monitoring/call recording capabilities. They are looking for an additional application, service or appliance that can sit on the LAN, and allow an administrator to monitor or recording inbound/outbound calls. If anyone is aware of a mechanism or solution that would provide this capability, please shoot me an email. Thanks Cory Andrews ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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] No CDR from Outbound Call
I have a little call recording script that I am running and it works
fine, but I have one problem. I get CDR when a user calls into the
extension, but I do not get CDR for the call that it makes outbound on #
17. Any idea why? Here is the extensions info:
[default]
exten = 2211,1,Answer
exten = 2211,2,Wait(1)
exten = 2211,3,Playback(/etc/asterisk/recording/getshop)
exten = 2211,4,playback(beep)
exten = 2211,5,Read(shopid)
exten = 2211,6,AGI,getnumber.agi|${shopid}
exten = 2211,7,Noop,${shopid}
exten = 2211,8,GotoIf($[${SHOPPHONE} = 1]?20:9)
exten = 2211,9,Noop,${SHOPPHONE}
exten = 2211,10,GotoIf($[${SHOPPHONE} = 2]?22:11)
exten = 2211,11,Noop,${SHOPNO}
exten = 2211,12,GotoIf($[${SHOPPHONE} = 3]?24:13)
exten = 2211,13,SetVar(CALLFILENAME=${SHOPNO}-${TIMESTAMP})
exten = 2211,14,AGI,startlog.agi|${SHOPPHONE}|${CALLFILENAME}
exten = 2211,15,SetCallerPres(prohib)
exten = 2211,15,SetCIDNum(2211)
exten = 2211,16,Monitor(wav,${CALLFILENAME},m)
exten = 2211,17,Dial(SIP/[EMAIL PROTECTED])
exten = 2211,18,wait(2)
exten = 2211,19,hangup
exten = 2211,20,playback(/etc/asterisk/recording/problem)
exten = 2211,21,goto(default,2211,2)
exten = 2211,22,playback(/etc/asterisk/recording/invalid)
exten = 2211,23,goto(default,2211,2)
exten = 2211,24,playback(/etc/asterisk/recording/syserror)
exten = 2211,25,goto(default,2211,2)
How it works is that a user calls in and enters a code. It then does a
database lookup of the code to find a number to call. It calls the
number and then bridges the two ends together, records it and mixes it
to an mp3. As I said, it works fine and has for over a year. The only
issue is that I don't get outbound CDR for some reason and I don't know
why
___
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[asterisk-users] Polycom Power Specs
Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --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] 2 devices using same sip account
It doesn't have anything to do with hardphone versus softphone. The issue is that it can only keep track of one registration per account. When the hardphone gets unplugged, it will not know about the softphone until it registers with asterisk. It's initial registration was lost when the hardphone registered with the same info. rilawich ango wrote: It seems that Greg is truth for the case. Asterisk doesn't care how many devices register to the same account as it is a feature of sip protocol (please let me know if there is a method to restrict it). In my case, I use a soft phone an hard phone using the same sip account information to register to the same asterisk. Soft phone register first and then hard phone register later. I dial the number and hard phone ring. Then I disconnect hard phone and expect soft phone will be ring after a couple of time. However, soft phone didn't ring as the call is failed. I issue database showkey SIP/Registry/sip account in CLI. It displays the information which belongs to hard phone. That's mean asterisk will keep the information of hard phone even it is disconnected with ignoring the soft phone registration. Does asterisk can be set to refresh its registry in a couple of time to remove the old registry record? On 12/19/06, Johansson Olle E [EMAIL PROTECTED] wrote: 19 dec 2006 kl. 11.58 skrev Gregory Duchatelet: Hi, It seems that they both can make calls, but only one can receive call: the last registered... Greg Hi all, What will happen if 2 devices using the same set of sip account to connect to the same asterisk? Do they both can make call? Can they receive call as normal? ___ --Bandwidth and Colocation provided by Easynews.com -- In Asterisk, you should only have one phone per account. We do not support multiple devices per account. The PBX core needs to know how many devices that we are calling each time we access it. /O ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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
Re: [asterisk-users] PRI to SIP
Virtually any Cisco device from a 2610 up will work. 2610, 2620, 2811, 2821, 3640, 3700, 3800. I have 2610 and 3640 in production for 2+ years with no issues. 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 -- Network stuff you didn't know http://www.networkoblivion.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] Low beep on voicemail
We've had a few people complain that the beep before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing about recording sounds, and I am sure I could spend a few hours and come up with a suitable version, but I thought I'd ask around before I waste my time trying to figure it out. Thanks in advance. Peder ___ --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, SQL ODBC
Is the storage of actual voicemail messages in a database still limited to ODBC? If so, why? And is the use of mySQL and ODBC at the same time still a bad idea? If so, why? I want to store all of my voicemail stuff in a database so that I can give users web access to it, but I don't want to run web services on my * server itself. If it is all in a DB, I can have a web box and a separate SQL box and none of it should affect *. ___ --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] Cisco GW CID Name
Does anybody know how to enable CallerID name passing from a Cisco gateway (with PRI that has name and number) to an * box via SIP? Supposedly CID name is enabled, but we can't get it passed to * and I've googled and I can't find what I need. ___ --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] Cisco PIX firewall and nat=yes
There is a Timeout SIP in the config. What is it set to? If it is less than the the qualify interval, which I believe is 60 seconds, then the PIX will close the inbound hole for qualify traffic. We've got lots of phones at several remote sites all running behind PIX's and all being NAT'd to the same IP (per location) and everything works perfect if qualify is on. If we disable qualify, then the SIP inbound hole gets closed per the Timeout SIP and calls don't go through until the phone re-registers and the hole opens again (they can still call out). Bill Gibbs wrote: As a follow up those commands helped with the outbound calls but inbound still had issues. Asterisk would still show the peer UNREACHABLE. Turning off qualify has fixed the problem! Bill *From:* Bill D'Anjou [mailto:[EMAIL PROTECTED] *Sent:* Wednesday, August 23, 2006 12:47 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* Bill Gibbs *Subject:* RE: [asterisk-users] Cisco PIX firewall and nat=yes You might need: fixup protocol sip 5060 fixup protocol sip udp 5060 in the PIX if these commands aren't supported you might need newer code. Bill -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Bill Gibbs *Sent:* Wednesday, August 23, 2006 8:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Cisco PIX firewall and nat=yes I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702x.x.x.x D N 54297UNREACHABLE 701/701x.x.x.x D N 54297UNREACHABLE 700/700x.x.x.x D N 54297UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because I’ve tested that out. So…I’m thinking it has something to do with the PIX. Any ideas? Bill ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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
Re: [asterisk-users] RTP Proxy
canreinvite=no will force all rtp packets through *. Ranjeet Kumar wrote: Hi, Can I do RTP Proxy in asterisk? As our requirement says that voice packet should also go though sip server, so that billing should be perfect. Thanks, Ranjeet Thanks, Ranjeet The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the individual or entity to whom they are addressed and is subject to legal privilege. If you have received this e-mail in error you should notify the sender immediately by reply e-mail, delete the message from your system and notify your system manager. Please do not copy it for any purpose, or disclose its contents to any other person. The views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of the company. The recipient should check this e-mail and any attachments for the presence of viruses. The company accepts no liability for any damage caused, directly or indirectly, by any virus transmitted in this email. www.aztecsoft.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 -- Network stuff you didn't know http://www.networkoblivion.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
Re: [asterisk-users] Max number of SIP devices registered to anextension
How does it work? Joshua Colp wrote: Rushowr wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Actually, isn't there SLA work being done in the trunk right now? It doesn't work how you think it does, you can still only have 1 SIP device registered to a SIP peer at a time. ___ --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] Uptime Record?
Our MOH died, so I finally had to kill my * process and restart it. Interestingly, stop now didn't work. I had to kill the process. It used to work, but it had been up so long that it must have gotten corrupted somehow. Here is the show uptime before I killed it: Asterisk-A*CLI show uptime System uptime: 1 year, 24 weeks, 3 days, 10 hours, 1 minute, 33 seconds Last reload: 11 hours, 30 minutes, 49 seconds Asterisk-A*CLI Who says * isn't stable enough for prime time? At least it is on 1.0.3. ___ --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] Uptime Record?
There aren't a lot of phones. There are 50-60 SIP phones and SIP connections to two Cisco PRI gateways. About 10,000 calls / month and about 15,000 mins of LD/month. I know when I started with *, I head how it had to be restarted every week and ours just ran and ran. Justin Tunney wrote: On 8/26/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Who says * isn't stable enough for prime time? At least it is on 1.0.3. What kind of abuse does that box take? ___ --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] Cisco PIX firewall and nat=yes
If you are running a new version of PIX sw (6.3.4 or 6.3.5), then leave fixup on and set nat=no. The PIX is the only firewall that I have seen that truly does nat correctly. It nat's both the source and dest inside the packet. You can even do reinvite with multiple phones behind a PIX and it works correctly. One other thing to check. If you have qualify off, then you need to set the phone to re-register in less time that the SIP timeout value in the PIX. For example, if the timeout is 10 mins, then the phone needs to have a register value less than 10 mins. Scott Pinhorne wrote: Hi I use a PIX 515 and had a similar problem when I started. I turned on the fixup for SIP (as well as having nat in sip entry) and it seems to do the trick for me. Good Luck SP Bill Gibbs wrote: Also the phone can dial out from behind the PIX…but obviously not receive calls. Bill *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Bill Gibbs *Sent:* Wednesday, August 23, 2006 11:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Cisco PIX firewall and nat=yes I have a Polycom 501 that works great from behind simple firewalls, like Dlink, etc however behind a Cisco PIX Firewall I see the register messages for the extensions on the Asterisk CLI but when I do a sip show peers I see: 702/702x.x.x.x D N 54297UNREACHABLE 701/701x.x.x.x D N 54297UNREACHABLE 700/700x.x.x.x D N 54297UNREACHABLE But I see stuff like n Registered SIP '702' at x.x.x.x port 54297 expires 60 I have a single phone with multiple extensions in the example above. As a test I changed that phone to a single extension (700), I see the Registered line but it still says UNREACHABLE. I know the Asterisk config is good because every device (soft, hard phone) works and I know the NAT works because I’ve tested that out. So…I’m thinking it has something to do with the PIX. Any ideas? Bill ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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
Re: [asterisk-users] t.38 bounty
What is the status of it anyway? I followed the bug for it and it appears that the bug was closed and maybe it was incorporated into Trunk. Is this true? And should it be (fully) functional now? PA ___ --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] Cisco Buddies
Can you do buddies with Cisco phones running SIP? I can't find anything that says yes or no. I can set it up on the * server, but I don't know what to do on the 7960's themselves. ___ --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] Cisco Buddies
I read both of those links and I don't see any mention of SIP buddies on either one. Adrià Vidal wrote: 2006/8/10, Peder @ NetworkOblivion [EMAIL PROTECTED]: Can you do buddies with Cisco phones running SIP? I can't find anything that says yes or no. I can set it up on the * server, but I don't know what to do on the 7960's themselves. What about a google look for asterisk cisco 7960 config in google? Firts and second looks great. Cisco 7960 IP Phone - SIP configuration - [ Traduzca esta página ] If you want to know how to configure your Cisco 7960 IP Phone to work with the skinny protocol (SCCP) and Asterisk PBX just click here. User Comments ... www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html - 33k - En caché - Páginas similares Asterisk phone cisco 79xx - voip-info.org - [ Traduzca esta página ] Asterisk Cisco 79XX XML Ser... Asterisk config sip.conf, Asterisk Linksys NSLU2, Cisco 7940-7960 auto-answer... cisco 79xx, Cisco Phones, Codecs ... www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx - 76k - -- Network stuff you didn't know http://www.networkoblivion.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
Re: [asterisk-users] Cisco Buddies
Then in the tftp config file for the phone add speeddials for the 6000 extension (cant recal how it is done, there are examples in the default file and on the wiki) I found out you really have to define the speeddials in the tftp files. speeddials configured with the phone menu or webinterface dont give you the status of the monitored phones. What tftp file? I've looked in SIPDefault and SIPmac and neither one has any mention of speed dials. I've searched all over the wiki and see no mention of buddies/presence on Cisco phones with SIP. ___ --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] ARA Regseconds
What is the regseconds field supposed to be used for when using ARA? I'm running 1.2.10 and when a phone registers, it is a HUGE number, like 1155074046. I assumed it would be the same as expire under a sip show peer, but it's not. That field shows 30 seconds (well it varies, but it's under 1 minute). ___ --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/Qualify
Are they both being NAT'd to the same external IP? Dovid Bender wrote: Hi List, I am not sure what this issue is. I am having a problem where I have 2 phones that are behind NAT on the same internet connection. The asterisk server has a public IP. Using asterisk real time1.2.10 on CentOS 4.3 with Ztdummy. For some reason I can only get ahold of one of the phones at a time. If I do sip show peers I get: Name/username HostDyn Nat ACL Port Status 10306/1030669.114.171.255 D N 8170 Unmonitored 10500/1050071.250.15.227D N 61249OK (134 ms) 10325/1032568.202.75.71 D N 2051 OK (91 ms) sipmedia/XX69.1.236.33 5060 Unmonitored 10310/1031069.114.171.255 D N 8174 OK (132 ms) 5 sip peers [5 online , 0 offline] I have Qualify=2000 (in mysql) but that dosent seem to work. I can only get to one of the phones but never both. Anyone have this before ? Dovid ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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
Re: [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102
When I looked several months ago, the only Sipura that supported T.38 was the SPA-2100. I haven't searched in a while, but I think it is still true. We go directly from a Cisco gateway to the SPA-2100 and it works great. It is the only ATA that we've seen that works right. Joshua Colp wrote: - Original Message - From: Olivier [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Wed, 26 Jul 2006 14:18:29 -0300 Subject: [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102 Hi, Which ATA supporting T.38 would you recommend (for reliability) ? Has anyone experienced this one ? http://www.voip-info.org/wiki/index.php?page=Linksys-Cisco+3102 Regards Hello There, I know that during our testing for T.38 capability in trunk Matt (the person who was doing the testing) went through the Grandstream ATAs initially and could not get them to work. Thanks to a generous donation he then moved onto trying with Sipura ATAs instead and they worked great so I would assume that the 3102 would also work nicely. Sipura (should I call them Linksys now?) have done a good job on their SIP stack and appear to have done a good job on their T.38 implementation too. If you do end up giving them a try, definitely report back so others will have some feedback. Have a great day! Joshua Colp Digium ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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
Re: [asterisk-users] Operator Console(s)/Shared Call Appearances
Does anybody know if shared appearance / BLA is on the * roadmap? And if so, when it might appear? I've seen people asking for it for quite a while, but I've never seen anybody say that it is in process or on the roadmap. ___ --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] G729 and Cisco IOS 12.4
Did anyone ever get an answer to this problem? I just brought a new gateway on line and it is running 12.4 and I have to use g729br8 and there are lots of quality issues with noise and errors on * about extra frames. If I drop the br8 codec, the phones can call out and the quality is great, but you can't call in from the gateway. Bill Gibbs wrote: Same thing...even with the commercial Digium G729 codec. I have to specifiy G729br8 on the Cisco. Cisco issue? Bill -Original Message- From: Bill Gibbs Sent: Monday, January 23, 2006 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] G729 and Cisco IOS 12.4 I have the same issue. I just bought the commercial version from Digium to see if that has the same problem. I wanted to use the free one to test out g729. My Polycom 301 had no issues using the free codec though (testing via VM, etc) Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Weiser Sent: Tuesday, December 20, 2005 12:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] G729 and Cisco IOS 12.4 Can anyone confirm that when using the G729 codec from http://kvin.lv/ pub/Linux/Asterisk/ and a Cisco gateway running IOS 12.4, codec negotiation fails? When I configure the dial-peer in the router with g729r8, it fails. If I use g729br8 (which uses a built-in VAD), it works. This behavior started since we upgraded the router from 12.3 (which had no issues). ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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] Cisco Gateway CallerID Name
I seem to remember reading somewhere about a setting on Cisco gateway's (with PRI) where you can have it send inbound (from PSTN) callerID name via SIP to *. Does anybody know what that setting is? I searched the archives and can't quite find the right set of keywords to locate that info. Thanks. Peder ___ --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] RE: Cisco Gateway CallerID Name
Two follow on questions: 1. Wouldn't that be for calls from * to the gateway out to the PSTN? I want incoming calls from the PSTN to the gateway to deliver CNAM via SIP to my * box. 2. How would I know if I want/need codeset 6? [EMAIL PROTECTED] wrote: In the interface Serial section add: isdn outgoing display-ie This will put the Display IE in codeset 0... if you need it in CodeSet 6 add: isdn outgoing ie display codeset_0 shiftcodeset codeset_6 Mark I seem to remember reading somewhere about a setting on Cisco gateway's (with PRI) where you can have it send inbound (from PSTN) callerID name via SIP to *. Does anybody know what that setting is? I searched the archives and can't quite find the right set of keywords to locate that info. Thanks. Peder ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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] Voicemail CallerID
I've got a question about voicemail and callerid and I can't quite
figure it out. I've got extensions 100, 101 and 102. For outbound
callerID (calls from the phones to the PSTN), I want the callerid to say
100 on all phones, so under sip.conf, I added:
callerid=Bill 100
The problem is that when they go to check voicemail, it looks at their
callerID and it drops them into mailbox 100 (calls to them still go into
their own specific mailbox, it is just when they hit their messages
button). Any idea how to get around that? Or do I just have to send
them to voicemail without having it automatically enter their extension?
This is what my voicemail does:
exten = 3299,1,VoicemailMain(${CALLERIDNUM})
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[asterisk-users] Problem - Can't pickup call
I've got a strange problem. I have two Cisco gateways each with one PRI and they each go to a different provider. One has been working for 2+ years with no problems. We recently added the second one and I have a problem where calls come in, but I can't answer them. The call comes into *, it rings the phone, but when you go to pick it up, it hangs up right away and starts ringing again. Here is the console output: -- Executing Dial(SIP/192.168.1.100-40431450, SIP/4230|30) in new stack -- Called 4230 -- SIP/4230-35bd is ringing -- SIP/4230-35bd answered SIP/192.168.1.100-40431450 -- Attempting native bridge of SIP/192.168.1.100-40431450 and SIP/4230-35bd == Spawn extension (CUSTOMER1, 4230, 1) exited non-zero on 'SIP/192.168.1.100-40431450' -- Executing Goto(SIP/192.168.1.100-40421ba8, CUSTOMER1|4230|1) in new stack -- Goto (CUSTOMER1,4230,1) -- Executing Dial(SIP/192.168.1.100-40421ba8, SIP/4230|30) in new stack -- Called 4230 -- SIP/4230-3734 is ringing -- SIP/4230-3734 answered SIP/192.168.1.100-40421ba8 -- Attempting native bridge of SIP/192.168.1.100-40421ba8 and SIP/4230-3734 == Spawn extension (CUSTOMER1, 4230, 1) exited non-zero on 'SIP/192.168.1.100-40421ba8' -- Executing Goto(SIP/192.168.1.100-40421ba8, CUSTOMER1|4230|1) in new stack -- Goto (CUSTOMER1,4230,1) -- Executing Dial(SIP/192.168.1.100-40421ba8, SIP/4230|30) in new stack -- Called 4230 -- SIP/4230-7714 is ringing -- SIP/4230-7714 answered SIP/192.168.1.100-40421ba8 -- Attempting native bridge of SIP/192.168.1.100-40421ba8 and SIP/4230-7714 == Spawn extension (CUSTOMER1, 4230, 1) exited non-zero on 'SIP/192.168.1.100-40421ba8' -- Executing Goto(SIP/192.168.1.100-40421ba8, CUSTOMER1|4230|1) in new stack -- Goto (CUSTOMER1,4230,1) etc. It dials, the phone rings, it attempts a bridge when you go off hook and then it is Executing Goto and starts ringing again. The strange thing is that I can call out that gateway and it works fine. It is only incoming calls that act strange. The config on the Cisco gateway is identical to the other one, but it is a different version of IOS. Here is the extensions.conf snippet for the two gateways (they are the exact same): [192.168.1.100] context=default type=friend host=192.168.1.100 dtmfmode=rfc2833 disallow=all allow=ulaw allow=g729 This is the extensions.conf part for the actual extension: exten = 4230,1,Dial(SIP/4230,30) exten = 4230,2,Voicemail(u4230) exten = 4230,102,Voicemail(b4230) exten = 4230,103,Hangup Any ideas? Or any ideas on how to figure out why it just keeps cycling over and over? ___ --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] Cisco SIP Firmware
What is the current recommended version of firmware for SIP on 7960/7940's. I was reading through some of the stuff on voip-info and it looks like the 8.x's have pretty serious bugs in regards ti *. Thanks. PA ___ --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 Contexts
Right now we have several companies all within on context in voicemail. The users want to have dial by name, so we are going to split into multiple contexts so that they don't accidentally dial each other (and complain). I've been reading over the voicemail context info and I'm somewhat confused and have the following question: Right now we have one main voicemail number that everybody uses. If they call internally, it doesn't matter what the number is because their phones dial it automatically. However, if they call from outside, they all use the same number that they already know. Once I split the contexts up, won't I have to have separate voicemail numbers for each voicemail context? My thought is that the system needs to know the context to check the mailbox because you could have the same 4 digit mailbox number in multiple contexts. Am I right that I need different numbers? If so, is that what everybody does? That seems like a waste of DID's. (we are running 1.0.3 and it's been running for System uptime: 1 year, 17 weeks, 1 day, 1 hour, 25 minutes, 6 seconds, so we aren't mucking with a new version if we don't have to) ___ --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] Cisco Buddies
Is there a buddies feature on the Cisco phones, like there is on the Polycom? If not, how are people getting around the issue where a receptionist wants to see who is on the phone? Or are they just living with the limitation? Thanks. Peder ___ --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 WAV to PDA Problems
Our asterisk server has been up and running for over a year and it works great. I have emails going to my account as an attachment and I can listen to them on the desktop and it works fine. I just got a T-Mobile MDA that runs Windows Pocket (or whatever they call it) and it can check email. If I have it download the email, it gets the attachment, but it can't seem to play it (it CAN play wav files). If I take the email that was sent to my home account and then forward it to myself and let the MDA pick it up, then it can play the attachment. So clearly it isn't an issue playing WAV's, or even WAV's from Asterisk, it's some email attachment issue with the way Asterisk or Postfix sends the attachment. Has anybody else run into this problem? If so, any help would be appreciated. Peder ___ --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] E911 from Remote Office via PRI
Not to be a smarta**, but you have to ask them to do it. We do the same thing and it works for us. Depending on the CLEC, they may do it or they may say no. If they say no, there isn't anything you can do about it. Hugh L. Johnson wrote: Central business location has a PRI with a CLEC. Remote offices access the PRI for all voice traffic via VoIP. How does one get the telco to report the address of a remote office to the 911 call center when the call is made from that respective location? ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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] 1.2 in production w/100+ phones?
Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime voicemail, etc), but I'm not sure if 1.2.1 is ready for primetime yet. Thanks. Peder -- Network stuff you didn't know http://www.networkoblivion.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
Re: [Asterisk-Users] SIP RTP
It just re-directs the RTP stream. The SIP stream still goes through *. Mike Hammett wrote: According to this page: http://www.asterisk.org/doxygen/Config_sip.html canreinvite=yes redirects just the RTP. I was under the impression that the entire SIP connection got redirected, therefore losing accounting ability. Could someone clarify this? --Mike ___ --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 -- Network stuff you didn't know http://www.networkoblivion.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] DaemonTools Supervise
Does anybody have a DaemonTools Supervise script for Asterisk? I searched google and the archives (and voip-info.org) and I see people mention using Supervise, but I don't see any actual sample scripts. Thanks. Peder ___ --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] SunFire X4100
Is anybody running * on a SunFire X4100? If so, any issues? Peder ___ --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
