[asterisk-users] Get return code from MeetMeAdmin()? did it possible?
?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal
Sent: Thursday, November 24, 2011 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MeetMeAdmin()
Hi,
I use this command exten =
unmute,n,MeetMeAdmin(room_number,m,${caller}),
Caller is the number of the participant that I want to mute, but when
the number of participant is not in the room I get a notice in the CLI:
NOTICE[17723]: app_meetme.c:4244 admin_exec: Specified User not found!
And the program continues from there as it should,
My question is, is there a way to use that answer in the program? Is
there a way to get a result of whether it works or not?
Thank you for your help.
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Re: [asterisk-users] A new hack?
On Sat, 26 Nov 2011, C F wrote: On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 26 Nov 2011, Terry Brummell wrote: Install Configure Fail2Ban then the host will be blocked from connecting. And no, it's not new. I don't need Fail2Ban, thank you. But your advice might be useful to others. Why is that? Even if they don't compromise an account they are still using your bandwidth and resources on your machine. Linux has excellent built-in subsystems to control firewalling and so on without resorting to external programs. It's called iptables. If you know how to use them, then using an external resource such as fail2ban is unneccessary. For example, with iptables rules you can say something like: If a connection from a remote site to a local port happens more than (say) once a second then drop that connection. And that happens right at the kernel level without the need to run any userland software, write config files, monitor log files and so on. I've posted about it in the past - search the archives if you want to know more. Gordon-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Asterisk alter the Headers of INVITE Message
Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email=m...@me.com) but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
Please guys anybody knows How can I send a unique token to the Receiver at the Invite call? Is that possible? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Sunday, November 27, 2011 11:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Does Asterisk alter the Headers of INVITE Message Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email=m...@me.com) but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
On 11/27/2011 04:27 PM, Faraj Khasib wrote: Please guys anybody knows How can I send a unique token to the Receiver at the Invite call? Is that possible? Custom SIP headers are a common way to do that. Try SIPAddHeader(). -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
I tried that with my SIP Cleint but the custom Header is not reaching the cleint ... Does the asketrisk delete that? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov [abalas...@evaristesys.com] Sent: Sunday, November 27, 2011 3:30 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message On 11/27/2011 04:27 PM, Faraj Khasib wrote: Please guys anybody knows How can I send a unique token to the Receiver at the Invite call? Is that possible? Custom SIP headers are a common way to do that. Try SIPAddHeader(). -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMS problems.
Hello, I tried to send sms for local extensions and i observed that file is created but sms isn't delivered yet. Can someone help me with this thing? rr:/var/spool/asterisk/sms/mttx # cat ../../outgoing/smsq.mttx.0.1322430026-20217.1 Channel: Local/1010 Callerid: SMS 1010 Application: SMS Data: 0,s MaxRetries: 0 RetryTime: 30 WaitTime: 10 rr:/var/spool/asterisk/sms/mttx # ls -la total 4 drwxr-xr-x 2 root root 88 Nov 27 23:40 . drwxr-xr-x 6 root root 144 Nov 27 23:40 .. -rw-r--r-- 1 root root 21 Nov 27 23:40 0.1322430026-20217 rr:/var/spool/asterisk/sms/mttx # cat 0.1322430026-20217 oa=1010 ud=TEST SMS. Thank you. P.S. I use smsq and asterisk 1.8.8.0. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
On 11/27/2011 04:53 PM, Faraj Khasib wrote: I tried that with my SIP Cleint but the custom Header is not reaching the cleint ... Does the asketrisk delete that? Are you sure? Have you taken a packet capture to confirm? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
You want to talk SIP, you need to talk SIP proxy. Hint: http://www.kamailio.org/w/ ;) Nick from Toronto. On Sun, Nov 27, 2011 at 5:19 PM, Alex Balashov abalas...@evaristesys.com wrote: On 11/27/2011 04:53 PM, Faraj Khasib wrote: I tried that with my SIP Cleint but the custom Header is not reaching the cleint ... Does the asketrisk delete that? Are you sure? Have you taken a packet capture to confirm? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
Yes, see attached ... Proxy server alter my Test custom header and delete it, Is there a way to include it in message sent from SIP Proxy to target? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov [abalas...@evaristesys.com] Sent: Sunday, November 27, 2011 4:19 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk alter the Headers ofINVITE Message On 11/27/2011 04:53 PM, Faraj Khasib wrote: I tried that with my SIP Cleint but the custom Header is not reaching the cleint ... Does the asketrisk delete that? Are you sure? Have you taken a packet capture to confirm? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Received (1128 bytes): 192.168.1.101:5060 - 192.168.1.104:50495 INVITE sip:6500@192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:50495;branch=z9hG4bK1421826827;rport From: sip:6097@192.168.1.101;tag=194243250 To: sip:6500@192.168.1.101 Contact: sip:6097@192.168.1.104:50495;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511 CSeq: 947428168 INVITE Content-Type: application/sdp Content-Length: 257 Max-Forwards: 70 Route: sip:192.168.1.101:5060;lr;transport=udp Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Test: testing Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel v=0 o=doubango 1983 678901 IN IP4 192.168.1.104 s=- c=IN IP4 192.168.1.104 t=0 0 m=audio 38378 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 Transaction(id='z9hG4bK1421826827' method=INVITE server=true) created. Sending (263 bytes): 192.168.1.101:5060 - 192.168.1.104:50495 begin SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:50495;branch=z9hG4bK1421826827;rport=50495 To: sip:6500@192.168.1.101 From: sip:6097@192.168.1.101;tag=194243250 Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511 CSeq: 947428168 INVITE Content-Length: 0 end Transaction(id='z9hG4bK1421826827' method=INVITE server=true) Transaction timeout Timer started, will triger after 9. Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE server=false) created. Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE server=false) Timer A(requst retransmit timer) started, will triger after 500. Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE server=false) Timer B(calling state timeout timer) started, will triger after 32000. Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE server=false) Transcation timeout timer started,timeout after 18 ms Sending (1199 bytes): 192.168.1.101:5060 - 192.168.1.101:59495 begin INVITE sip:6500@192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101;branch=z9hG4bK-d32f84d469d6407daffba73dccb7cadb;rport Via: SIP/2.0/UDP 192.168.1.104:50495;branch=z9hG4bK1421826827;rport=50495 From: sip:6097@192.168.1.101;tag=194243250 To: sip:6500@192.168.1.101 Contact: sip:6097@192.168.1.104:50495;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511 CSeq: 947428168 INVITE Content-Type: application/sdp Max-Forwards: 69 Accept-Contact: *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Test: testing Allow: INVITE,ACK,CANCEL,BYE,MESSAGE,OPTIONS,NOTIFY,PRACK,UPDATE,REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel Record-Route: sip:192.168.1.101;lr Content-Length: 257 v=0 o=doubango 1983 678901 IN IP4 192.168.1.104 s=- c=IN IP4 192.168.1.104 t=0 0 m=audio 38378 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 end Received (421 bytes): 192.168.1.101:5060 - 192.168.1.101:59495 SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/UDP 192.168.1.101;rport;branch=z9hG4bK-d32f84d469d6407daffba73dccb7cadb From: sip:6097@192.168.1.101;tag=194243250 To:
[asterisk-users] Displaying entered digits in the LCD of the IP Phone when is requested to enter it
Hi All; When using the IP Phone and will be prompted to enter a digits (for example, using the Background function), so when dialing the digits, I do not see these digits on the LCD of the Phone, how can I make it appear? Actually I need it because I have a Background funtion that ask the agent to enter the number (through some scenario), so when dialing it, we would this to appear at the LCD of the phone, how? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
On 11/27/2011 05:25 PM, Faraj Khasib wrote: Yes, see attached ... Proxy server alter my Test custom header and delete it, Is there a way to include it in message sent from SIP Proxy to target? That would be a proxy configuration issue, wouldn't it? In principle, the proxy should be passing these messages through unmodified, unless you have an explicit configuration directive that instructs it to remove headers from the INVITE. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
thats my main question if u can see Does Asterisk alter the Headersof INVITE Message I am using ASterisk NOW proxy I didnt configure it to delete anything , Can u tell me how I can change it to pass that parameters? thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov [abalas...@evaristesys.com] Sent: Sunday, November 27, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message On 11/27/2011 05:25 PM, Faraj Khasib wrote: Yes, see attached ... Proxy server alter my Test custom header and delete it, Is there a way to include it in message sent from SIP Proxy to target? That would be a proxy configuration issue, wouldn't it? In principle, the proxy should be passing these messages through unmodified, unless you have an explicit configuration directive that instructs it to remove headers from the INVITE. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
Any body knows how I can configure Asterisk SIP to pass all Header Parameters? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Sunday, November 27, 2011 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message thats my main question if u can see Does Asterisk alter the Headersof INVITE Message I am using ASterisk NOW proxy I didnt configure it to delete anything , Can u tell me how I can change it to pass that parameters? thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov [abalas...@evaristesys.com] Sent: Sunday, November 27, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message On 11/27/2011 05:25 PM, Faraj Khasib wrote: Yes, see attached ... Proxy server alter my Test custom header and delete it, Is there a way to include it in message sent from SIP Proxy to target? That would be a proxy configuration issue, wouldn't it? In principle, the proxy should be passing these messages through unmodified, unless you have an explicit configuration directive that instructs it to remove headers from the INVITE. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?
Hi All, While I'm certainly comfortable compiling from sources, I'm trying to do an rpm only asterisk install on CentOS 5.7. I'm using the asterisk repositories and I installed all the asterisk18 rpms, but find that chan_gtalk and res_jabber are missing. Is there a separate rpm that includes support for gtalk? Thanks in advance. -Gaurav -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?
It has been almost a year since I suggested to consider including these into the RPM build. There was no friction ever since, and I am building from sources too... It seems the RPM maintainers think that Google Voice connectivity is an experimental feature and thus it should not be included in the RPM. Or maybe their logic is different. The end result is the same. -Vladimir On 11/27/2011 7:22 PM, Gaurav P wrote: Hi All, While I'm certainly comfortable compiling from sources, I'm trying to do an rpm only asterisk install on CentOS 5.7. I'm using the asterisk repositories and I installed all the asterisk18 rpms, but find that chan_gtalk and res_jabber are missing. Is there a separate rpm that includes support for gtalk? Thanks in advance. -Gaurav -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?
Do you build from source and copy res_jabber.so and chan_gtalk.so to the rpm installed directories? Or have you just given up on the packages and instead build from source? On Sun, Nov 27, 2011 at 8:35 PM, Vladimir Mikhelson v...@mikhelson.comwrote: It has been almost a year since I suggested to consider including these into the RPM build. There was no friction ever since, and I am building from sources too... It seems the RPM maintainers think that Google Voice connectivity is an experimental feature and thus it should not be included in the RPM. Or maybe their logic is different. The end result is the same. -Vladimir On 11/27/2011 7:22 PM, Gaurav P wrote: Hi All, While I'm certainly comfortable compiling from sources, I'm trying to do an rpm only asterisk install on CentOS 5.7. I'm using the asterisk repositories and I installed all the asterisk18 rpms, but find that chan_gtalk and res_jabber are missing. Is there a separate rpm that includes support for gtalk? Thanks in advance. -Gaurav -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?
I just go through the whole process. * ./configure * make menu * make * make install I tried building pieces but then ran into the problem where Asterisk was not happy with different version of my modules. I tried to inquire what specific flags or other parameters I needed to use while compiling so that my modules would be accepted into the RPM delivered Asterisk, but ran into the same wall of silence. As I maintain all other components by yum update I need to install Asterisk and then overwrite it by make install. The same with DAHDI. Not very elegant or convenient. -Vladimir On 11/27/2011 10:23 PM, Gaurav P wrote: Do you build from source and copy res_jabber.so and chan_gtalk.so to the rpm installed directories? Or have you just given up on the packages and instead build from source? On Sun, Nov 27, 2011 at 8:35 PM, Vladimir Mikhelson v...@mikhelson.com mailto:v...@mikhelson.com wrote: It has been almost a year since I suggested to consider including these into the RPM build. There was no friction ever since, and I am building from sources too... It seems the RPM maintainers think that Google Voice connectivity is an experimental feature and thus it should not be included in the RPM. Or maybe their logic is different. The end result is the same. -Vladimir On 11/27/2011 7:22 PM, Gaurav P wrote: Hi All, While I'm certainly comfortable compiling from sources, I'm trying to do an rpm only asterisk install on CentOS 5.7. I'm using the asterisk repositories and I installed all the asterisk18 rpms, but find that chan_gtalk and res_jabber are missing. Is there a separate rpm that includes support for gtalk? Thanks in advance. -Gaurav -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
Well, It doesn't forward the INVITE at all, as asterisk is NOT a proxy. It
creates a totally new INVITE when you issue the Dial application, with its
own set of headers.
Now, you can pass the Test header with something like this (taken from
memory...):
SipAddHeader(Test: ${SIP_HEADER(Test)})
Do that prior to the call to the Dial application, and you will see your
header in the outgoing INVITE. Of course this means that your dial plan need
to know which headers to pass.
// T
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: den 28 november 2011 00:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE
Message
Any body knows how I can configure Asterisk SIP to pass all Header
Parameters?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Sunday, November 27, 2011 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Does Asterisk alter the Headers of
INVITE Message
thats my main question if u can see Does Asterisk alter the Headersof
INVITE Message
I am using ASterisk NOW proxy I didnt configure it to delete anything ,
Can u tell me how I can change it to pass that parameters?
thanx
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
[abalas...@evaristesys.com]
Sent: Sunday, November 27, 2011 4:41 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk alter the Headers of
INVITE Message
On 11/27/2011 05:25 PM, Faraj Khasib wrote:
Yes, see attached ... Proxy server alter my Test custom header and
delete it, Is there a way to include it in message sent from SIP
Proxy to target?
That would be a proxy configuration issue, wouldn't it?
In principle, the proxy should be passing these messages through
unmodified, unless you have an explicit configuration directive that
instructs it to remove headers from the INVITE.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
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