[asterisk-users] Question about Cisco IP phone + Asterisk + channels
Hi all, I'm planning on picking up a Cisco IP phone or 2 and have a question about the multiple lines feature of them, and Asterisk channels in general. Lets say I have 2 Cisco IP phones and a call comes in, each one rings line 1, and I pick up. Is there any way to have notification on the other phone that I'm currently on that channel? If so, then what about if a 2nd call comes in, will it automatically start ringing the 2nd line on the phone? I've never played around with these phones except at my wife's college dorm, which wasn't much. Basically right now I have some ATAs with cordless phones hooked up to them and each ATA has it's own line sort to speak (I'm sure you guys know what I mean), where as one call comes in and whoever answers first wins the channel. If anyone is confused by this and needs clarification, let me know. Thanks in advance! :) ___ -- 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] More Broadvoice woes. Who's fault could this be?
Forwarding isn't on. If I call from my cell: fax machine. Call from a land line: call gets through. Call from Skype: fax machine again! If only phone numbers were trace routable like IP addressed, to see where the heck my calls are going. On Mon, Mar 24, 2008 at 12:49 AM, John Faubion [EMAIL PROTECTED] wrote: cell phone. When I do, I get a fax macine. Debugging SIP shows NO call activity what so ever. Make sure you don't have it forwarded to another number at BV. John ___ -- 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] More Broadvoice woes. Who's fault could thisbe?
I did do an upgrade to 1.4 (from 1.2), which is what lead me to try phoning my BV number from my cell. It's long distance for me so I don't call it ever. I wasn't able to get calls so I was placing a few test calls to see if I could get it to ring. I was getting fast busy signals at first but then a few days later I get the fax machine sound when I try to call. Oddly enough though, as I sit here typing this email, I decided to try a softphone to see what happens.. Called using my Link2VoIP account, and yet a 3rd source that I got the fax machine noise from. Decided I would give BV a call, but for whatever reason I reconnected it back to Asterisk and give it ONE last shot, and the damn thing works now. 100% works. I think I'm still going to port away from them though. Their customer service really is sub-standard. Thanks all for your help though :) On Mon, Mar 24, 2008 at 11:37 AM, Outback Dingo [EMAIL PROTECTED] wrote: have you modified your configs lately or rebooted your box lately. i mean im asking what if anything has changed On Mon, Mar 24, 2008 at 10:05 PM, John Faubion [EMAIL PROTECTED] wrote: Forwarding isn'tn. If I call from my cell: fax machine. Call from a land line: call gets through. Call from Skype: fax machine again! Ok now that is bizarre. Three different sources and two different destinations. Looks like the only common point would be BV. Sounds like your going to get to spend some time on the phone with BV. If only phone numbers were trace routable like IP addressed, to see where the heck my calls are going. I'd really like to see that feature as well! John ___ -- 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] More Broadvoice woes. Who's fault could this be?
Hi all, I'm not sure if this is the correct mailing list for this (I was going to send to Asterisk-Biz, but seems more for this one). Anyway, I'm having more problems with Broadvoice. I still can't get calls unless I comment out the secret= line in sip.conf, but now I can't even place test calls to it from my cell phone. When I do, I get a fax macine. Debugging SIP shows NO call activity what so ever. This call clearly isn't being directed to my BV number. If I try to call it using Skype I get the same thing. Calling from a land line I don't the fax machine, but rather a fast busy signal because my Asterisk box is rejecting the call. I'm really a loss at who to contact. I have a feeling if I phone my cell phone provider they'll blow me off, and I know Broadvoice won't even answer my emails. Has anyone else ever had this happen? I'm in the process of porting the number over to another company. Any idea if this problem might follow me once that's completed? Thanks in advance! ___ -- 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] Problem with incoming calls on Broadvoice after upgrade to 1.4.18
Hi Raj, Sorry for the delay. The NIC in my server running Asterisk died so I wasn't able to verify until just now. After commenting out the secret= line, calls go through. I'll contact their support, but I'm sure they'll be as useless as ever. This may be the last straw for them. Thanks again Raj On Sun, Mar 16, 2008 at 6:44 PM, Raj Jain [EMAIL PROTECTED] wrote: Based on the trace alone, it seems like a problem on their end. You may want to try shutting off INVITE authentication (by commenting out secret= line in your sip.conf) to see if the call goes through. On Sun, Mar 16, 2008 at 6:27 PM, Jon Miron [EMAIL PROTECTED] wrote: Hi Raj, Thanks for your response. I'm a little confused though. Does this look as if it's a problem with Broadvoice itself, and not my configuration? Any time I've called them with problems where it's clearly not my fault (ie nothing on my end has changed), they're never very helpful. On Sun, Mar 16, 2008 at 4:45 PM, Raj Jain [EMAIL PROTECTED] wrote: Looking at the trace, the entity sending you the INVITE is not resubmitting INVITE with credentials after the initial INVITE was challenged with a 401 response by Asterisk. The trace shows two independent calls and both have the same problem. -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org On Sun, Mar 16, 2008 at 4:10 PM, Jon Miron [EMAIL PROTECTED] wrote: Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable debugging I get the following: SIP Debugging Enabled for IP: 147.135.0.128 net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- INVITE sip:my Broadvoice #@servers IP:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=prtu To: my namesip:s@servers IP Via: SIP/2.0/UDP 147.135.0.128:5060 Contact: sip:my cell #@147.135.0.128:5060 Supported: 100rel Content-Length: 309 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.250 t=0 0 m=audio 28274 RTP/AVP 0 8 18 96 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:97 t38/8000 a=rtpmap:101 telephone-event/8000 - --- (10 headers 14 lines) --- == Using SIP RTP CoS mark 5 Sending to 147.135.0.128 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] No user 'my cell #' in SIP users list Found peer 'sip.broadvoice.com' for 'my cell #' from 147.135.0.128:5060 net-xero*CLI --- Reliably Transmitting (no NAT) to 147.135.0.128:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128 From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=prtu To: my namesip:s@servers IP;tag=as77a74c13 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r106946 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=06b61489 Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- ACK sip:my Broadvoice #@servers IP:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=prtu To: my namesip:s@servers IP;tag=as77a74c13 Via: SIP/2.0/UDP 147.135.0.128:5060 Content-Length:0 - --- (7 headers 0 lines) --- [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:9020 sip_reregister: -- Re-registration for my Broadvoice #@sip.broadvoice.com REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 147.135.0.128:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP servers IP:5060;branch=z9hG4bK339bbe4e;rport Max-Forwards: 70 From: sip:my Broadvoice #@sip.broadvoice.com;tag
[asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18
Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable debugging I get the following: SIP Debugging Enabled for IP: 147.135.0.128 net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- INVITE sip:my Broadvoice #@servers IP:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=prtu To: my namesip:s@servers IP Via: SIP/2.0/UDP 147.135.0.128:5060 Contact: sip:my cell #@147.135.0.128:5060 Supported: 100rel Content-Length: 309 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.250 t=0 0 m=audio 28274 RTP/AVP 0 8 18 96 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:97 t38/8000 a=rtpmap:101 telephone-event/8000 - --- (10 headers 14 lines) --- == Using SIP RTP CoS mark 5 Sending to 147.135.0.128 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] No user 'my cell #' in SIP users list Found peer 'sip.broadvoice.com' for 'my cell #' from 147.135.0.128:5060 net-xero*CLI --- Reliably Transmitting (no NAT) to 147.135.0.128:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128 From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=prtu To: my namesip:s@servers IP;tag=as77a74c13 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r106946 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=06b61489 Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- ACK sip:my Broadvoice #@servers IP:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=prtu To: my namesip:s@servers IP;tag=as77a74c13 Via: SIP/2.0/UDP 147.135.0.128:5060 Content-Length:0 - --- (7 headers 0 lines) --- [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:9020 sip_reregister:-- Re-registration for my Broadvoice #@sip.broadvoice.com REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 147.135.0.128:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP servers IP:5060;branch=z9hG4bK339bbe4e;rport Max-Forwards: 70 From: sip:my Broadvoice #@sip.broadvoice.com;tag=as2a457f50 To: sip:my Broadvoice #@sip.broadvoice.com Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX SVN-trunk-r106946 Expires: 120 Contact: sip:s@servers IP Event: registration Content-Length: 0 --- net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER From: sip:my Broadvoice #@sip.broadvoice.com;tag=as2a457f50 To: sip:my Broadvoice #@sip.broadvoice.com Via: SIP/2.0/UDP servers IP:5060;branch=z9hG4bK339bbe4e Contact: sip:s@servers IP Expires: 30 Event: registration Content-Length:0 - --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:14949 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- INVITE sip:my Broadvoice #@servers IP:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=ikmn To: my namesip:s@servers IP Via: SIP/2.0/UDP 147.135.0.128:5060 Contact: sip:my cell #@147.135.0.128:5060 Supported: 100rel Content-Length: 309 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.250 t=0 0 m=audio 28276 RTP/AVP 0 8 18 96 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:97 t38/8000 a=rtpmap:101 telephone-event/8000 - --- (10 headers 14 lines) --- == Using SIP RTP CoS mark 5 Sending to 147.135.0.128 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] No user 'my cell #' in SIP users list Found peer 'sip.broadvoice.com' for 'my cell #' from 147.135.0.128:5060 net-xero*CLI --- Reliably Transmitting (no NAT) to 147.135.0.128:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128 From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=ikmn To: my namesip:s@servers IP;tag=as6ef11459 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX
Re: [asterisk-users] Problem with incoming calls on Broadvoice after upgrade to 1.4.18
Hi Raj, Thanks for your response. I'm a little confused though. Does this look as if it's a problem with Broadvoice itself, and not my configuration? Any time I've called them with problems where it's clearly not my fault (ie nothing on my end has changed), they're never very helpful. On Sun, Mar 16, 2008 at 4:45 PM, Raj Jain [EMAIL PROTECTED] wrote: Looking at the trace, the entity sending you the INVITE is not resubmitting INVITE with credentials after the initial INVITE was challenged with a 401 response by Asterisk. The trace shows two independent calls and both have the same problem. -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org On Sun, Mar 16, 2008 at 4:10 PM, Jon Miron [EMAIL PROTECTED] wrote: Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable debugging I get the following: SIP Debugging Enabled for IP: 147.135.0.128 net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- INVITE sip:my Broadvoice #@servers IP:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=prtu To: my namesip:s@servers IP Via: SIP/2.0/UDP 147.135.0.128:5060 Contact: sip:my cell #@147.135.0.128:5060 Supported: 100rel Content-Length: 309 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.250 t=0 0 m=audio 28274 RTP/AVP 0 8 18 96 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:97 t38/8000 a=rtpmap:101 telephone-event/8000 - --- (10 headers 14 lines) --- == Using SIP RTP CoS mark 5 Sending to 147.135.0.128 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] No user 'my cell #' in SIP users list Found peer 'sip.broadvoice.com' for 'my cell #' from 147.135.0.128:5060 net-xero*CLI --- Reliably Transmitting (no NAT) to 147.135.0.128:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.0.128:5060;received=147.135.0.128 From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=prtu To: my namesip:s@servers IP;tag=as77a74c13 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-trunk-r106946 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=06b61489 Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- ACK sip:my Broadvoice #@servers IP:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: Toronto ONsip:my cell #@147.135.0.128;user=phone;tag=prtu To: my namesip:s@servers IP;tag=as77a74c13 Via: SIP/2.0/UDP 147.135.0.128:5060 Content-Length:0 - --- (7 headers 0 lines) --- [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:9020 sip_reregister:-- Re-registration for my Broadvoice #@sip.broadvoice.com REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 147.135.0.128:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP servers IP:5060;branch=z9hG4bK339bbe4e;rport Max-Forwards: 70 From: sip:my Broadvoice #@sip.broadvoice.com;tag=as2a457f50 To: sip:my Broadvoice #@sip.broadvoice.com Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX SVN-trunk-r106946 Expires: 120 Contact: sip:s@servers IP Event: registration Content-Length: 0 --- net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER From: sip:my Broadvoice #@sip.broadvoice.com;tag=as2a457f50 To: sip:my Broadvoice #@sip.broadvoice.com Via: SIP/2.0/UDP servers IP:5060;branch=z9hG4bK339bbe4e Contact: sip:s@servers IP Expires: 30 Event: registration Content-Length:0 - --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) [Mar 16 15:52:39] NOTICE[27524]: chan_sip.c:14949 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) net-xero*CLI --- SIP read from UDP://147.135.0.128:5060 --- INVITE sip:my Broadvoice
[Asterisk-Users] Limiting the number of calls
Hey everyone. I'm wondering if anyone has any ideas on a way to limit the number of outbound calls at a time, and if the limit is reached a message is played when someone tries to place the next call. I've searched the wiki but have yet to come up with anything. Any help would be appreciated. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BroadVoice patch on latest CVS snapshot
Hey, I'm trying to patch the latest CVS snapshot with the BroadVoice patch but I get this when I try: [EMAIL PROTECTED]:/usr/src/asterisk/channels# patch chan_sip.c sip_patch.diff patching file chan_sip.c Hunk #1 FAILED at 213. Hunk #2 succeeded at 315 (offset 9 lines). Hunk #3 FAILED at 485. Hunk #4 FAILED at 493. Hunk #5 succeeded at 3986 (offset 20 lines). Hunk #6 succeeded at 3996 (offset 9 lines). Hunk #7 succeeded at 4027 (offset 20 lines). Hunk #8 succeeded at 4051 with fuzz 2 (offset 12 lines). Hunk #9 succeeded at 4101 (offset 21 lines). Hunk #10 succeeded at 4108 (offset 12 lines). Hunk #11 succeeded at 4147 (offset 21 lines). Hunk #12 succeeded at 4153 (offset 12 lines). Hunk #13 succeeded at 4276 (offset 21 lines). Hunk #14 succeeded at 6271 (offset 56 lines). Hunk #15 succeeded at 6360 (offset 21 lines). Hunk #16 succeeded at 6810 (offset 58 lines). Hunk #17 FAILED at 6844. 4 out of 17 hunks FAILED -- saving rejects to file chan_sip.c.rej Any ideas on what I can do? Giving my box an external IP and avoiding the patch completetly is an option, but I'd rather leave it NATed. Thanks in advance! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfering incoming calls using same line
Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + phone# and have it do a flash then have it dial *08 the same phone number + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would like to be able to make long distance calls to any number while I'm out with my cell phone so I want to take advantage of my free LD package on my PSTN line. Thanks in advance! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfering incoming calls using same line
--- Shaun Ewing [EMAIL PROTECTED] wrote:
On Wed, 22 Sep 2004 10:20:58 -0400 (EDT), Jon Miron
[EMAIL PROTECTED] wrote:
Hey all,
Wondering if this is possible.. Incoming call is
answered through X100P, then an extension is
dialed
using the same X100P card. Basically I want to
dial
in, enter 9 + phone# and have it do a flash then
have it dial *08 the same phone number + # on
the
same PSTN line to have it transfer my call to
another
phone number. I realize this isn't very safe, but
I
would like to be able to make long distance calls
to
any number while I'm out with my cell phone so I
want
to take advantage of my free LD package on my PSTN
line. Thanks in advance!
Three applications that would allow you to achieve
something like that
- disa, flash and senddtmf.
This is untested, but some logic like the following
might help:
Use something like the following in your IVR.
exten = 10,1,DISA,1234|calltransfer
Then, add the context and code like:
[calltransfer]
exten = _X.,1,Flash
exten = _X.,2,Wait,1
exten = _X.,3,SendDTMF(*08${EXTEN}#)
exten = _X.,4,Hangup
I'm not sure off hand how you could terminate a
number with #, but
DISA will proceed after 10 seconds anyway.
I'd be curious to see if it works - so let us know.
Actually the # key tells my service provider that I'm
finished with the call and want it to be handed off.
For example I'm talking to person #1 and they want to
talk to person #2, so I flash over, do *08 + person
#2's number, and once they pick up I'd be like Person
#1 wants to speak to you, hold on and press # then
hang up and the two are connected and my line goes
silent.
Thanks for the dialplan. I'll let you know how it
works when I get my X100P installed early next week.
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[Asterisk-Users] Setting up Asterisk with fwd
Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register = xx:[EMAIL PROTECTED]/489125 [fwd.pulver.com] type=friend secret=xxx username=xx fromuser=xx fromdomain=fwd.pulver.com host=fwd.pulver.com dtmfmode=inband nat=yes canreinvite=novv Is there something wrong with this? Why isn't it signing into fwd? Also, I wanted to set it up so I could dial 8 + fwd # and have it call out using this, but it doesn't appear to work. Apparently this Asterisk stuff is harder than it looks :) exten = _8.,2,Dial(SIP/[EMAIL PROTECTED]/$(EXTEN),20,tr) Thanks in advance for any help! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What would be required for this?
--- Greg Hill [EMAIL PROTECTED] wrote:
This is relatively straightforward to implement in a
dialplan
(extensions.conf) either by implementing extensions
direction or by using
the DISA application. Keep in mind that a system
which allows an incoming
call to make an outgoing call has some inherent
security issues
('wardialing' hasn't gone away).
Is there any way to add some kind of secuity to this?
Either by needing an access code to get into some kind
of sub-menu, only allow it from certain numbers (eg my
cell number), or some other way?
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[Asterisk-Users] What would be required for this?
Hey All, I have a question that I'm curious about. I want to set up a 4 phone system in my home with 2 actual lines coming into the house. Both or just regular lines (not sure of this matters?), one being VoIP and the other just a regular analog line. For now though I just want the VoIP line coming in, but would like the ability to expand to 2 lines in the future. What type of hardware is required for this, and how much would it cost? For now though, this is what I want to do and for as cheap as possible.. I have a VoIP line that has free long distance on it and I want to be able to dial into Astrisk from my cell to be able to reach any number I want (eg extention that dials an outside line). Any ideas on how to go about this? Thanks in advance! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Primus Talkbroadband
Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense. I'm completely new to this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Primus Talkbroadband
Jon, Does Primus actually use MGCP though? I've heard mix results (though keep in mind I only became interested in all of this earlier today, so I know very little). I checked the specs on my dlink and it says it's SIPs with no mention of MGCP. However everywhere else says Primus is not SIPs. As far as quality, I've had mixed results as well. One of my co-workers says it sounds horrible, but no one else seems to notice anything. Could be the fact that I'm using a cordless phone though. Incoming call quality seems pretty good. Even when downloading at 500K/sec+ it still seems to sound pretty good with only minor choppiness. My best suggestion though, is to get it for a month or so and try it out seeing as it's only $20/month. --- Jon Pounder [EMAIL PROTECTED] wrote: I have asked before and got no answers - I am still not clear as to why there is not an MGCP client as part of asterisk - is it a technical reason, no one else wants it, other ? It is my understanding primus is using mgcp, and therefore is unable to directly interface with asterisk, password or not. I would like to hear your experiences with quality though since I am considering switching to it anyway and run the dlink boxes into my channel bank for now, and figure out the software issues later. Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense. I'm completely new to this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Primus Talkbroadband
Jon, Hmm I didn't know about the versions thing. I'll have to get the exact model number off the device when i get home. I never set up any forwarding at all for it though. I simply plugged it into my switch and everything was up and running within a few seconds. Not sure if that's a good sign or not, but probably in the end it will turn out Primus is actually MGCP. Like I said, I haven't played around with this much at all, but is there at least a little support for MGCP in Asterisk? There's a /etc/asterisk/mgcp.conf file, and when you run the Asterisk console it has a bunch of mgcp commands. Or am I mistaken? --- Jon Pounder [EMAIL PROTECTED] wrote: What is the exact model of your gateway ? There is an M and S model that are very similar in looks/features except one is sip and one is mgcp (at least last time I looked in detail at what they were supplying) What ports did you open in your firewall for this to work ? that should be another way to tell unless you are wide open. I would try it but I really don't want yet another number, I have numbers in Mississauga and St.Catharines I would like to migrate as long as it is decent quality. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Primus Talkbroadband
Geoff, How frequent are your dropped calls? For a while all my calls would go silent but I realized it was after exactly 60 minutes. It's since been increased to 180. Not sure if this is what you were experiencing. Are there any providers in Canada that offer a similar service to Primus that is more Asterisk friendly? I just need it to have numbers in Toronto, and Montreal if possible. Thanks! --- Geoff Nordli [EMAIL PROTECTED] wrote: I did a packet sniff and it is definitely MGCP. I find that the quality hasn't been great. I am looking at moving to something different. It frequently drops calls, but I don't know if it is the NAT device that does it. I would like to find something that is a little bit more Asterisk friendly. I just need to find a provider that can give me DIDs in various BC locations. I have the 1120M/PR model. Have a great day! Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
