Re: [asterisk-users] Realtime peers and trunks coming from the same IP
Thanks Sammy, I think I'll stop using SIP realtime. Regards, Ricardo. On Mon, May 21, 2012 at 5:14 AM, SamyGo govoi...@gmail.com wrote: Hello Ricardo, The reason why your asterisk refused the calls from phone registering on SIP proxy is that it only gets INVITE of the call from: a user that is defined BUT Not Registered within asterisk. The easy way of solving this is 1- Stop asterisk SIP realtime and let only the SIP proxy handle registrations. 2- Tell asterisk to accept calls from the SIP proxy only (create a SIP peer for proxy) This will make everything work. Regards, Sammy. On Sat, May 19, 2012 at 9:15 PM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: I use an SBC to protect my pool of asterisk servers and as trunking endpoint with SIP Telcos. Now I'm trying to implement SIP phone registration to be delegated through the SBC, as a proxy. It doesn't work. It just works when I don't use realtime peers at the asterisk servers. Using realtime SIP peers, since there is one SIP phone that gets his registration delegated through the SBC, any inbound call that tries to reach any asterisk server, coming from any SIP Telco trunk ended at my SBC, gets refused in asterisk. As asterisk records the IP of the SBC as the IP of the phone that has been registered, it thinks that those calls coming from the SBC are calls coming from that phone, and it refuses them with 401 Unauthorized replies. I'm using asterisk 1.8.11. How can I surpass this problem? Is there any configuration that I'm lacking on, or is this a limitation of asterisk? Thanks, Ricardo. -- _ -- 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] Realtime peers and trunks coming from the same IP
I use an SBC to protect my pool of asterisk servers and as trunking endpoint with SIP Telcos. Now I'm trying to implement SIP phone registration to be delegated through the SBC, as a proxy. It doesn't work. It just works when I don't use realtime peers at the asterisk servers. Using realtime SIP peers, since there is one SIP phone that gets his registration delegated through the SBC, any inbound call that tries to reach any asterisk server, coming from any SIP Telco trunk ended at my SBC, gets refused in asterisk. As asterisk records the IP of the SBC as the IP of the phone that has been registered, it thinks that those calls coming from the SBC are calls coming from that phone, and it refuses them with 401 Unauthorized replies. I'm using asterisk 1.8.11. How can I surpass this problem? Is there any configuration that I'm lacking on, or is this a limitation of asterisk? Thanks, Ricardo. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore. I don't know why!... This is the SDP portion that comes in the INVITE messages of calls through that trunk (let's say, whose endpoint has the IP x.x.x.x, purposely omitted). Nothing seems to be wrong with that to me: v=0 o=CSM 0 1 IN IP4 x.x.x.x s=Acme c=IN IP4 x.x.x.x t=0 0 m=audio 22152 RTP/AVP 8 0 18 4 101 a=rtpmap:101 telephone-event/8000 And here's the debugging: [May 8 17:45:30] DEBUG[6444]: chan_sip.c:5092 do_setnat: Setting NAT on RTP to Off [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP o=CSM 0 1 IN IP4 x.x.x.x... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP s=Acme... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting 'x.x.x.x' into... [May 8 17:45:30] DEBUG[6444]: netsock2.c:188 ast_sockaddr_split_hostport: ...host 'x.x.x.x' and port ''. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP c=IN IP4 x.x.x.x... OK. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 4 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: chan_sip.c:9110 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 4 on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x416e25b0 [May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x416e25b0 [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Thanks, Ricardo. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
Problem SOLVED. You'r right, this is a problem of codec mismatching. Activating sip debug i can see it: Capabilities: us - 0x802 (gsm|g726), peer - audio=0x10d (g723|ulaw|alaw|g729) [May 9 17:16:37] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! I solved the problem thanks to your help! Since that SIP trunk isn't authenticated, i just receive calls in the default context that is set in sip.conf, and so, I don't set the codecs to be used. I discovered that the problem was that i had one other peer defined in sip.conf that had the same IP address set, so it was shuffling asterisk some how. Funny that the same configuration wasn't a problem in asterisk 1.4, but in this 1.8 it caused this problem. Thank you onde again, Regards, Ricardo. On Wed, May 9, 2012 at 5:10 PM, Andres and...@telesip.net wrote: On 5/9/2012 11:56 AM, Ricardo Carvalho wrote: That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. My guess is the incoming call is not being matched with the peer you are expecting. Do a sip debug and watch the output to see what peer is being selected. Andres Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.ukmailto: asterisk_l...@earthshod.co.uk wrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] any enum test number of e164.arpa tree?
Can anybody please tell me any ENUM test DID from e164.arpa tree, which I can use to test some features? Thanks, Ricardo. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk security....again
Probably, you are receiving INVITE attacks from some tool like sipvicious. You should rearange your network to cover some inportant security issues. The IP address of you server can be revealed in some unincrypted SIP signaling of some call through the Internet to/from your server's client, or simply by your client SRV record in the DNS, if you added it to his DNS. Probably your network is exposed to the Internet. To address those situations, you can use a distinct VLAN to address SIP phones and you also can use port security at the switching ports where you connect your ATAs and phones. You should also deliver with tagging (802.1Q) that VLAN to those ATAs and phones. This should protect you from inside sniffers. This VLAN should just communicate with the DMZ where you should have your asterisk server and between those two networks you should only open the needed ports - for a common SIP infrastructure you should open UDP 5060 and the specified UDP range shown in rtp.conf file for the media to pass. Phones VLAN should not communicate directlly with the world, just in the outbound direction if you like. Regards, Ricardo Carvalho. On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham rizwanhas...@gmail.comwrote: Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with Asterisk Unknown caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server is thru sip-uri calls directly to customers. I have updated the customers atas to not accept any calls from sources other than the registration server. Thats all fine now. But the question is how can anyone know the direct sip uri addresses of our customers. My guess is that someone has been sniffing my server's sip traffic. In that case what should i do to get rid of the sniffers? If you think there is another reason for that then please tell me even if you dont have the solution. Thanks -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.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] trunk not working if I register a phone at the same IP as the trunk peer's IP
Thanks Faisal, in fact I made a test that confirmed that in realtime asterisk doesn’t supported static peers, like you told me. Do you know if newer versions of asterisk, like 1.8, have this issue already solved? Regards, Ricardo. On Wed, Feb 16, 2011 at 6:26 PM, Faisal Hanif fai...@vopium.com wrote: I have played a lot on this issue with asterisk config but in realtime it doesn’t supported static peers with version 1.6.2.14. *From:* Ricardo Carvalho [mailto:rjcarvalho.li...@gmail.com] *Sent:* Wednesday, February 16, 2011 10:21 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* Faisal Hanif *Subject:* Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP Isn't this a limitation that can be surpassed with some configuration that I'm lacking in my sip.conf or extensions.conf of my asterisk? Ricardo. On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif fai...@vopium.com wrote: Well a quick n easy fix for you is you can configure you call sending peers to use username secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs. If you can’t do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds. Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho *Sent:* Wednesday, February 16, 2011 9:41 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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] trunk not working if I register a phone at the same IP as the trunk peer's IP
How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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] trunk not working if I register a phone at the same IP as the trunk peer's IP
Isn't this a limitation that can be surpassed with some configuration that I'm lacking in my sip.conf or extensions.conf of my asterisk? Ricardo. On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif fai...@vopium.com wrote: Well a quick n easy fix for you is you can configure you call sending peers to use username secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs. If you can’t do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds. Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho *Sent:* Wednesday, February 16, 2011 9:41 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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] trunks and phones registered from the same IP
I sent a few hours earlier another e-mail to this list detailing a bit more my problem. Please see it with, it has the following subject: unregistered trunks and registered phones coming from the same IP Thanks, Ricardo. On Tue, Feb 15, 2011 at 12:36 PM, Pezhman Lali l...@lopl.net wrote: really it's too difficult to understand, please explain more clear On Tue, Feb 15, 2011 at 5:17 AM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: Hi, How can I configure my asterisk server so that I can receive incomming calls comming from the same IP from where my server also receives phone registrations? The problem is that since the moment any extension registers at that IP (actually I have a registration proxy running at that IP), asterisk no more accepts calls coming from a SIP trunk I also have at that IP, replying back with 401 Unauthorized. Any ideas? Thanks, Ricardo. -- _ -- 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] unregistered trunks and registered phones coming from the same IP
At the SBC, I delegate the registration to my asterisk server forwarding the REGISTER requests to it and resetting the contact to itself. This should allow my asterisk server to forward subsequent messages to the SBC rather than to the phone client directly. When the SBC receives a 200 OK from the asterisk registrar for the delegated REGISTER request, it saves the binding into its location database. I am almost sure that the configuration at the SBC is right, the problem seems to be related with the asterisk behaviour, as it refuses with 407 Proxy Authentication Required replay (if I use asterisk 1.4, or with 401 Unauthorized if I use asterisk 1.8) those calls coming from the SBC as a trusted trunk for which I don't require authentication. Asterisk seems to think that those calls came from the phone whose registration has proxied at the SBC, and only accepts inbound calls from the SBC if its INVITEs carry that phone correct authentication. Any ideas? Thanks, Ricardo. On Tue, Feb 15, 2011 at 12:50 PM, Faisal Hanif fai...@vopium.com wrote: You need to use relay request in your SBC instead of forward. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pezhman Lali *Sent:* Tuesday, February 15, 2011 5:42 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] unregistered trunks and registered phones coming from the same IP please send your sip.conf, is any NAT procedure implemented in your network? On Mon, Feb 14, 2011 at 10:16 PM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: Hi, I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound and outbound calls. Now I've configured the SBC to also act as a registration proxy, forwarding SIP registrations coming from the Internet to my asterisk servers. It all seems fine, but it doesn't work well, because by the time at least one phone registers through the SBC to some asterisk server (lets say, server_A), future incoming calls coming from my SIP telco trunks to my server_A got refused by the asterisk running on that server, with 401 Unauthorized messages back to the SBC. Seems like that since the moment asterisk binds some contact to the IP of the SBC, because it registered through it, from that moment, asterisk only accepts calls from that IP if those INVITEs carry correct registration to my server (even if those calls came from my SBC, a trusted trunk, not registered in asterisk). My phones are configured with type=friend. I've also tried type=peer and type=user, but it doesn't solve the problem. Any ideas to fix this? Best regards, Ricardo. -- _ -- 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] unregistered trunks and registered phones coming from the same IP
Hi, I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound and outbound calls. Now I've configured the SBC to also act as a registration proxy, forwarding SIP registrations coming from the Internet to my asterisk servers. It all seems fine, but it doesn't work well, because by the time at least one phone registers through the SBC to some asterisk server (lets say, server_A), future incoming calls coming from my SIP telco trunks to my server_A got refused by the asterisk running on that server, with 401 Unauthorized messages back to the SBC. Seems like that since the moment asterisk binds some contact to the IP of the SBC, because it registered through it, from that moment, asterisk only accepts calls from that IP if those INVITEs carry correct registration to my server (even if those calls came from my SBC, a trusted trunk, not registered in asterisk). My phones are configured with type=friend. I've also tried type=peer and type=user, but it doesn't solve the problem. Any ideas to fix this? Best regards, Ricardo. -- _ -- 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] trunks and phones registered from the same IP
Hi, How can I configure my asterisk server so that I can receive incomming calls comming from the same IP from where my server also receives phone registrations? The problem is that since the moment any extension registers at that IP (actually I have a registration proxy running at that IP), asterisk no more accepts calls coming from a SIP trunk I also have at that IP, replying back with 401 Unauthorized. Any ideas? Thanks, Ricardo. -- _ -- 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] sip.conf outboundproxy
So, does anyone ever used outboundproxy in sip.conf with success? Does it only send OUTBOUND calls via the proxy and not also internal extension calls via that proxy? Best Regards, Ricardo. ___ -- 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] sip.conf outboundproxy
Thanks Kevin. Although it doesn't fit my needs, thanks for the explanation. I guess I'll really have to combine Asterisk with OpenSer to do what I want. Ricardo. On Thu, Mar 26, 2009 at 1:07 PM, Kevin P. Fleming kpflem...@digium.comwrote: Ricardo Carvalho wrote: Does it only send OUTBOUND calls via the proxy and not also internal extension calls via that proxy? As has already been posted in your other threads about this subject, Asterisk has no concept of an 'outbound' call at all. In that sense, the name of this option in sip.conf is incorrect, it should just be 'proxy'. If you tell Asterisk to use a SIP proxy for sending out SIP requests, it will send all requests to that proxy, regardless of whether that request might be involved in a call that you classify as 'internal'. To Asterisk, a SIP call is a SIP call; there is no 'internal', 'external', 'outbound', 'inbound', at least not in the sense of 'inside my PBX' or 'outside my PBX'. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] sip.conf outboundproxy
The problem is that I cannot put the outboundproxy statement to the applicable sip extension context, due to the fact that I want to force every ENUM call to go via the proxy; and ENUM calls don't use any context to leave asterisk. Even so, putting outboundproxy statement is in the global section of sip.conf, for internal calls destined to phones registered in the same asterisk server, I think asterisk should see those are internal calls and don't ship the signaling through the proxy, right? Ricardo. On Tue, Mar 24, 2009 at 7:08 PM, Danny Nicholas da...@debsinc.com wrote: Just a guess, but your outboundproxy statement is in the global section of sip.conf, which is making it apply to all sip traffic. If you move that line to the applicable sip extension (ie. prox...@sipprov.com), this will probably fix the behavior, even if it doesn't resolve the problem. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Tuesday, March 24, 2009 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip.conf outboundproxy On 24 Mar 2009, at 17:51, Ricardo Carvalho wrote: Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Never used it before, but in the mind of Asterisk, how is your sip handset any different to a provider? Its outbound from asterisk.. I may be wrong.. Steve ___ -- 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] sip.conf outboundproxy
Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Am I doing something wrong or is this the real behaviour of the outboundproxy variable in sip.conf? Best Regards, Ricardo Carvalho. ___ -- 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] T.38 VoIP providers
By your experience, please someone tell me which T.38 capable VoIP SIP providers have you tested with success sending and receiving FAX with Asterisk 1.4. Thanks, Ricardo Carvalho. ___ -- 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] Calling users to the external domain using Asterisk
What you are looking for is something like this piece of code. Adapt it for
your scenario:
[default]
exten = _.,1,NoOp(incomming call from ${CALLERID} to [EMAIL PROTECTED])
exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _.,6,GotoIf($[${SIPDOMAIN} = 192.168.1.1]?10)
exten = _.,7,NoOp(@${SIPDOMAIN} is from an external domain, sending to
it...)
exten = _.,8,Dial(SIP/[EMAIL PROTECTED])
exten = _.,9,HangUp()
exten = _.,10,Goto(noturi-default,${EXTEN},1)
exten = h,1,HangUp()
[noturi-default]
;(your dialplan)
Regards,
Ricardo Carvalho.
On Thu, Mar 27, 2008 at 7:47 AM, Aadilkhan Maniyar [EMAIL PROTECTED]
wrote:
Hi All,
I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and
using it to make SIP calls.
I have a configuration of Asterisk which serves the users in a particular
domain, say internal.com
I would like to make a SIP call from [EMAIL PROTECTED] to
[EMAIL PROTECTED]
I have added the following lines in extensions.conf
exten = charles,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]
)
exten = charles,2,Hangup
Asterisk does a DNS SRV lookup and resolves the external.com to its proper
IP and calls are established.
But the problem with the above configuration is that I have manually added
users that are in the external domain.
Is there any way wherein I can call the users in external.com without
adding them in the extensions.conf?
Any help would be appreciated.
Thanks,
Aadil
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Re: [asterisk-users] Calling users to the external domain usingAsterisk
You can test manually any SRV DNS record using dig, like this:
dig -t SRV _sip._udp.fwd.pulver.com
At the asterisk CLI you can also verify that SRV lookup has been succeeded.
It shows something like this when it does:
parse_srv: SRV mapped to host fwd.pulver.com, port 5060
In your dialplan you can also trigger some Set(CDR(userfield)=SRV call from
${SIPCHANINFO(recvip)}) so that in your mysql CDR table be written which
calls got sent by IP to any SIP URI.
Regards,
Ricardo Carvalho.
On Fri, Mar 28, 2008 at 12:00 PM, Aadilkhan Maniyar [EMAIL PROTECTED]
wrote:
Thanks for the reply Recardo..
I was indeed looking at something like this…
Also I was also looking at Asterisk's SRV lookups… Is there anyway I can
know that a SRV lookup has failed?
Regards,
Aadil
-Original Message-
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Ricardo Carvalho
*Sent:* Friday, March 28, 2008 4:07 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Calling users to the external domain
usingAsterisk
What you are looking for is something like this piece of code. Adapt it
for your scenario:
[default]
exten = _.,1,NoOp(incomming call from ${CALLERID} to
[EMAIL PROTECTED])
exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _.,6,GotoIf($[${SIPDOMAIN} = 192.168.1.1]?10)
exten = _.,7,NoOp(@${SIPDOMAIN} is from an external domain, sending to
it...)
exten = _.,8,Dial(SIP/[EMAIL PROTECTED])
exten = _.,9,HangUp()
exten = _.,10,Goto(noturi-default,${EXTEN},1)
exten = h,1,HangUp()
[noturi-default]
;(your dialplan)
Regards,
Ricardo Carvalho.
On Thu, Mar 27, 2008 at 7:47 AM, Aadilkhan Maniyar
[EMAIL PROTECTED] wrote:
Hi All,
I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and
using it to make SIP calls.
I have a configuration of Asterisk which serves the users in a particular
domain, say internal.com
I would like to make a SIP call from [EMAIL PROTECTED] to
[EMAIL PROTECTED]
I have added the following lines in extensions.conf
exten = charles,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]
)
exten = charles,2,Hangup
Asterisk does a DNS SRV lookup and resolves the external.com to its proper
IP and calls are established.
But the problem with the above configuration is that I have manually added
users that are in the external domain.
Is there any way wherein I can call the users in external.com without
adding them in the extensions.conf?
Any help would be appreciated.
Thanks,
Aadil
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Re: [asterisk-users] Realtime replication!!!!!
Take it: http://www.onlamp.com/pub/a/onlamp/2006/04/20/advanced-mysql-replication.html?page=1 Regards, Ricardo Carvalho. On Wed, Mar 26, 2008 at 10:45 AM, Al Baker [EMAIL PROTECTED] wrote: Could you point a link to the DUAL MASTER Replication. I swear i have been all over the docs and have NOT found this, thx ___ -- 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] T.38 SIP Issues
I made some tests with FAX in Asterisk 1.4 using T.38 between two ATAs connected to legacy FAX machines, and realized that only SIP can make passthrough in the server while RTP go direct between endpoints. Is it possible for RTP data stream also to make passthrough in Asterisk? Thanks, Ricardo Carvalho. On Fri, Mar 14, 2008 at 1:13 PM, Steve Underwood [EMAIL PROTECTED] wrote: Mindaugas Kezys wrote: Hello, Higher speeds then 9600kbps are not permited by patents. Would you care to name one that prevents 14,400? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing Solution for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Friday, March 14, 2008 3:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T.38 SIP Issues Has someone submitted a bugreport regarding enabling 9600kbps fax? I always wonder why it would never negociate 14400kbps... when it did work a single page on fine resolution would take 4 minutes. Thank you very much for that link. I knew there had to be more possible configurations for T.38. I will give it a try... but I think I can get away without patching chan_sip.c, no? that just seems to enable higher bitrates. And Linksys SPA2102 is one of the exact devices I have in my lab. On Thu, Mar 13, 2008 at 1:52 PM, Mindaugas Kezys [EMAIL PROTECTED] wrote: Hello, This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Thursday, March 13, 2008 5:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 SIP Issues Is there any trick to getting T.38 fax to work with SIP? I had it working and one day with no changes *poof* it stopped working and hasn't worked for months. The only common factor is Asterisk 1.4.x (always try to use the latest version) and NAT. I've tried: -Linksys ATA -Grandstream ATA -Audicodes ATA All do the same thing. Call connects, hear the first 2sec of fax tone and then just silence, but the call usually stays open. I've tried two T.38-capable providers. I've tried two different routers: -Linksys WRT54GS running DD-WRT (Linux) -Dell Optiplex 170L running PFSense (BSD) Different Linux distros on the servers: -SuSE 64bit -RHEL 32bit -SuSE 32bit Is there any magic to get this to work? As far as I can tell the only possible config option is t38pt_udptl = yes which I have set under [general] the peer. Steve ___ -- 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] T.38 passthrough in asterisk 1.4
I've been testing the exchange of FAXes in Asterisk 1.4 using T.38, and I found that it only works if I use canreinvite=yes for both ATAs connected to my legacy FAX machines. The so called T.38 passthrough is only for SIP signaling? Not for RTP? If so, is there a way I can make the RTP traffic of those FAXes go out using the same IP of my Asterisk server through the SIP trunk I have established with the telco I have subscribed? This is because, this telco only accepts all traffic (SIP+RTP) sent by the same IP. Regards, Ricardo Carvalho. ___ -- 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] two lines written in CDR for each failed call in asterisk 1.4
I've upgraded my server from asterisk 1.2 to 1.4.18 and CDR has started to log two lines for each failed call (NO ANSWER) instead of one per call as it was done in 1.2. In my extensions.conf, call flow starts in default context and then jumps to a macro where is then dialed the destination. When call isn't answered CDR logs two lines, one with the s extension in dst and the outbound channel in channel which is related with the macro; and other line with the correct destination number in dst and the incoming channel in channel. Is this behavior expected in Asterisk 1.4 or didn't I port correctly my dialplan syntax from 1.2 to 1.4? Since I want CDR to be written like it was in 1.2, can this feature be disabled somehow? Regards, Ricardo Carvalho. ___ -- 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] logging the estimated RTT using SIP
Is it possible in Asterisk 1.4 to log by somehow the estimated roundtrip time (RTT) between server and some peer, which Asterisk computes based on the sending of OPTIONS and the receiving of the responses to those OPTIONS? Regards, Ricardo Carvalho. ___ -- 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] SNMP monitoring
I had the same problem some time ago... You got to install also this packages: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel That should do it! Regards, Ricardo Carvalho. On Thu, Feb 14, 2008 at 5:30 PM, Adrian Marsh [EMAIL PROTECTED] wrote: Hi All, I've been reading up on 1.4 snmp integration. When I try and compile asterisk with a –with-netsnmp option it complains about net-snmp installation being broken. However, the net-snmp-devel rpm is installed, and snmpd on the machine runs fine. Anyone have a guide for the pre-requisites needed ? Cheers, Adrian ___ -- 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] SNMP monitoring
Maybe you'r right and newt isn't really necessary. I just read somewhere that those dependencies were needed, I've installed them and it worked... Try to only install the other ones and if res_snmp gets compiled without it, great! Regards, Ricardo Carvalho. On Fri, Feb 15, 2008 at 12:01 AM, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Ricardo Carvalho wrote: I had the same problem some time ago... You got to install also this packages: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel That should do it! Why would this depend on newt? net-snmp and lm-sensor headers and libraries make sense. newt doesn't make any sense as a dependency. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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
Re: [asterisk-users] Asterisk and fax
I'm at this moment implementing the same as you do... Take a look at the following links: http://blog.evaristesys.com/?p=24 http://blogtech.oc9.com/index.php?option=com_contentview=articlecatid=4:asteriskid=77:20071121astItemid=6 http://www.voip-info.org/wiki/view/Asterisk+fax Regards, Ricardo Carvalho. On Feb 13, 2008 5:49 PM, voip crazy [EMAIL PROTECTED] wrote: I want to receibe the fax via mail and send faxes via web interface and a digital send and receibe fax list. Voipcrazy 2008/2/13, Giorgio Incantalupo [EMAIL PROTECTED]: Hi VoIPCrazy, why don't you use an ATA device such as Grandstream 486 or similar? Giorgio Incantalupo voip crazy wrote: Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1 mode. Which is the best solution to use with this hardware? Which solution do you use to send an receibe fax? Thanks VoIPCrazy ___ -- 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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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] Logging in and off sessions in the dialplan
Is it possible to implement in the Asterisk dialplan some way to authenticate a user with a dialed passcode which opens session that stays active enabling the user to make and receive calls, until the user logs off with another dialed passcode? I am aware of the Asterisk application 'Authenticate', but as far as I know, with this application the user meeds to dial his pin at each call he whats to make, and that not what I need! Some ideas? Thanks, Ricardo Carvalho. ___ --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] Cisco power injector with GXP2000 phones
I've tried to use a Cisco power injector to supply power over Ethernet to a GXP2000 phone without success. Although when I plugged these phone to a PoE capable Cisco Switch it worked without a problem! Knowing that all these three equipments implement IEEE 802.3af protocol, why doesn't it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho. ___ --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] Logging in and off sessions in the dialplan
Thanks Gordon, I'll give it a try with astDB. Regards, Ricardo Carvalho. ___ --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 power injector with GXP2000 phones
I only see one explanation to my problem... GXP2000 phones only implement PoE mode A of the IEEE 802.3af protocol, and the power injector does only PoE mode B of the IEEE 802.3af protocol. The switch does mode A. The problem is that I can't prove this! can't find documentation with this kind of detail. If someone does, please tell. Regards, Ricardo Carvalho. ___ --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] Do While loop
You can try something like this:
exten = _X.,1,SET(condition=${RAND(1,2)})
exten = _X.,2,GotoIf($[${condition} = '1']?1:3)
exten = _X.,3,SET(Result is 2)
Regards,
Ricardo Carvalho.
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Re: [asterisk-users] Shared line appearance phones?
I don't know if I understood you right, but can't that be solved with call queues? http://www.voip-info.org/wiki/index.php?page=Asterisk+call+queues http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf Regards, Ricardo Carvalho ___ --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 behind a PIX firewall?
Try to just open port 5060 for SIP signaling on the PIX and also enable the INSPECT SIP rule. That way, your PIX firewall will inspect SIP signalling and open the necessary UDP ports for the RTP. If you have NAT uptream in the network, you should see if in the layer 4 the IPs shown in the SIP messages got rewritten by its public IPs, it should have, or else you'll never get it working right. Regards, Ricardo Carvalho. ___ --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] Finding the status of an extension
If you have some SIP phone BLF feature capable, you can try it. With it in the phone you can view the state of those the registered extensions you like, as well with it if you do sip show subscriptions in the asterisk CLI, you'll get the list of extensions and the last state of those monitored extensions. Regards, Ricardo Carvalho. ___ --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] hostname in MySQL CDR records
You can also use CDR(userfield) parameter and that way you can write in the column userfield of your CDR table of the DB, the hostname of the asterisk for each call. You can try something like the following in the dialplan of each machine: Set(CDR(userfield)=hostname_of_the_machine) Regards, Ricardo Carvalho. ___ --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] Get IP address of an incoming or outgoing SIP call
You may take a look at the SIPCHANINFO(recvip) function. With it, you can even start logging into CDR the IPs of incoming and outgoing calls. Regards, Ricardo Carvalho. ___ --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] Softphone to be installed on the Mobile
Here's one sip softphone for mobiles you can give a try: http://www.minisip.org/ Regards, Ricardo Carvalho. ___ --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 extensions configuration - calling user filtering
As much I as can tell, Asterisk version 1.2 doesn't support the ex-girlfriend logic that you ask. I didn't test that feature with 1.4 releases, maybe they already implement it. Regards, Ricardo Carvalho.. On Nov 20, 2007 2:51 PM, Tomasz Zieleniewski [EMAIL PROTECTED] wrote: Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten = some_exten/calling-user is there some flag which activates this extra check?? Cheers Tomasz ___ --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] Realtime extensions configuration - calling user filtering
If you really want to use some DB to read/write your dialplan, the best thing for you would be to write some scripts to generate text files from the contents of the tables of your DB. Those files can then be loaded in the extensions.conf file with the sentence: #include generated_file.txt. In the same script you can even do some asterisk -r -x extensions reload command, and then you'll have your own realtime extensions working with the ex-girlfriend logic you wanted! I implemented this way because I had the same problem as you... :) Regards, Ricardo Carvalho. On Nov 20, 2007 4:16 PM, Tomasz Zieleniewski [EMAIL PROTECTED] wrote: I tried it with 1.4 and it didn't work with standard settings and no magic:) On Nov 20, 2007 4:32 PM, Ricardo Carvalho [EMAIL PROTECTED] wrote: As much I as can tell, Asterisk version 1.2 doesn't support the ex-girlfriend logic that you ask. I didn't test that feature with 1.4 releases, maybe they already implement it. Regards, Ricardo Carvalho.. On Nov 20, 2007 2:51 PM, Tomasz Zieleniewski [EMAIL PROTECTED] wrote: Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten = some_exten/calling-user is there some flag which activates this extra check?? Cheers Tomasz ___ --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 GXP2020 / 2000
We've a site with about 200 Grandstream GXP2000 phones, and they work quite well. We made some CGI Perl scripts to mass-deploy and manage their configurations from a MySQL DB into a TFTP server, where the phones go to download their binaries. With some initial work, now it has become easy to manage the site. All phones have firmware version 1.1.1.14; we are testing new stable version 1.1.4.18 but by now we found that some phones freeze sometimes - version 1.1.1.14 seems more stable. One thing they lack is the ability to dial alphanumeric contacts (URI dials), we hope future firmware corrects this issue. Older ones hadn't so much good hands-free speaker, but recent ones have a better DSP from Texas Instruments. Althow they're not the best choice in the market (like Cisco or Polycom), they represent a good price/quality ratio. Regards, Ricardo Carvalho. ___ 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] faster timeout in ENUMLOOKUP() function
Hi all, In my server dialplan, it first tries to dial possible SIP URI contacts returned by DNS lookups using the ENUMLOOKUP function; it only sends calls to PSTN when there aren't any NAPTR records of the dialed number. Problem arises when my Internet connection is down to some locations, which inhibits my Asterisk server to reach the DNS servers to do those lookups. In those cases, calls only get sent to the PSTN after ENUMLOOKUP function times out (which takes very long)! Is it possible to configure a shorter timeout for the ENUMLOOKUP function, so the next priority in my dialplan comes faster? Or any ideas to avoid this problem? Regards, Ricardo Carvalho. ___ 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] Authenticate() application and CDR
Dear all,
I'm trying to configure Asterisk to be able to ask the caller to enter a
given password in order to continue dialplan execution. I've tested this
feature using the Authenticate application like this:
exten = _X./5219,1,Answer
exten = _X./5219,2,Authenticate(1234,a)
exten = _X./5219,3,Playback(pin-number-accepted)
exten = _X./5219,4,Dial(SIP/${EXTEN},120)
Works great, although there is a problem with the CDR: Asterisk accounts
only the call that is answered by Asterisk which asks for the pin. Case
the call hasn't been answered by the called party, or case it even
hasn't been dialed because the caller failed to insert the pin, or even
if it has been answered, Asterisk writes in CDR table that it has been
ANSWERED and billed from the time Asterisk picked up to ask the pin.
I'm I skipping something in my syntax, or is this some kind of BUG? (I'm
using Asterisk version 1.2.17)
Regards,
--
Ricardo Carvalho
ITEC / IRICUP / Reitoria UP
tel: +351220408108 (Ext: 5219)
e-mail/sip: rjcarvalho[at]reit.up.pt
--
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Re: [asterisk-users] Authenticate() application and CDR
Thanks Atis, You've helped a lot. Regards, Ricardo. -- Ricardo Carvalho ITEC / IRICUP / Reitoria UP tel: +351220408108 (Ext: 5219) e-mail/sip: rjcarvalho[at]reit.up.pt -- Atis Lezdins wrote: On Friday 21 September 2007 19:07:43 Ricardo Carvalho wrote: Works great, although there is a problem with the CDR: Asterisk accounts only the call that is answered by Asterisk which asks for the pin. Case the call hasn't been answered by the called party, or case it even hasn't been dialed because the caller failed to insert the pin, or even if it has been answered, Asterisk writes in CDR table that it has been ANSWERED and billed from the time Asterisk picked up to ask the pin. I'm I skipping something in my syntax, or is this some kind of BUG? (I'm using Asterisk version 1.2.17) Nop, your dialplan is correct, and this is not a bug. Answer() in first line marks incoming call answered, so counter (also from your provider) is on, and you can't turn it off. Of course, Answer() is required, so that asterisk can start receiving voice, and DTMF to authenticate. If you would want to do your own billing, to count only duration of call dialed to SIP/whatever, you can do 1,Answer() 2,Authenticate() 3,Playback() 4,ResetCDR() 5,Dial() NoCDR would tell to not write CDR for that channel, but ResetCDR later would reset answer status for CDR, and start counting duration from that moment. ResetCDR(w) would make you have two CDR records, one for each part (that can be linked together by using uniqueid). Regards, Atis ___ 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] how to load phone registration information
I'm using realtime sip already! To let you understant better my problem, I'll explain a bit more: In a redundancy scheme, I have two asterisk servers, each running on different machines although sharing the same MySQL DB for relatime sip. Problem arises when the second server assumes the production. When some phone tries to establish a new call, those INVITEs reach the new server, although this server seems to don't read the registration information kept in sip_buddies table to know if the destination phone is registered or not, and so, the call fails. Because the destination phone was registered in the first server, I was expecting that the second server when assuming production would first read the sip_buddies DB table to see if the destination phone was registered or not, but that seems to don't happen. It seems that registration information is only kept in memory and isn't read from DB! Is there any way that I can force Asterisk to read sip_buddies realtime DB table to know if destination phone is registered? Regards, Ricardo Carvalho. On 7/12/07, Ricardo Carvalho [EMAIL PROTECTED] wrote: Is it possible to load phone registration information stored in sipfriends MySQL DB, so that Asterisk thinks those phones are already registered? This would be very usefull for a redundant server... Look at realtime sip should help you ram ___ --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] how to load phone registration information
Is it possible to load phone registration information stored in sipfriends MySQL DB, so that Asterisk thinks those phones are already registered? This would be very usefull for a redundant server... Regards, Ricardo Carvalho. ___ --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] how to force asterisk to read registration information from DB
In a redundancy scheme, I have two asterisk servers, each running on different machines although sharing the same MySQL DB for relatime sip. Problem arises when the second server assumes the production. When some phone tries to establish a new call, those INVITEs reach the new server, although this server seems to don't read the registration information kept in sip_buddies table to know if the destination phone is registered or not, and so, the call fails. Because the destination phone was registered in the first server, I was expecting that the second server when assuming production would first read the sip_buddies DB table to see if the destination phone was registered or not, but that seems to don't happen. It seems that registration information is only kept in memory and isn't read from DB! Is there any way that I can force Asterisk to read sip_buddies realtime DB table to know if destination phone is registered? Regards, Ricardo Carvalho. ___ --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] sharing phone registration information between asterisk servers
Is it possible to share SIP phones registration information between two different asterisk servers, that share the same realtime MySQL DB? Regards, Ricardo. ___ --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] Forcing Dial application to skip if called server is unreachable
In fact, Dial() doesn't return instantly like it should, in the case it
is used with ENUM. Dial application using the ENUMLOOKUP function
doesn't skip to the next priority like it was expected, if destination
server doesn't answer to the INVITE messages sent by our server.
For example, in the following code, if the first Dial using ENUM fails
to reach the contact's server, instead of skipping to the next priority
Dialing Zap channel instead, Asterisk keeps sending INVITE messages to
the destination server published in ENUM until dial timeout expires
(120), and only then jumps to the next priority, Dialing Zap:
exten = _X.,1,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0)
exten = _X.,2,GotoIf($[${counter}${sipcount}]?3:6)
exten = _X.,3,Set(counter=$[${counter}+1])
exten = _X.,4,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter})})
exten = _X.,5,GotoIf($[${counter}${sipcount}]?3:6)
exten = _X.,6,Dial(Zap/g1/${EXTEN})
Is this an Asterisk BUG or is it there some way I can solve this problem?
Regards,
Ricardo.
Alex Balashov wrote:
On Wed, 20 Jun 2007, [EMAIL PROTECTED] wrote:
Is it possible to force the Dial function to skip to the next priority if it
doesn't find the server of the called contact within a few seconds?
I know I can use:
Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL])
where I can use some short timeout in the timeout option, but if I do so,
when some call is well succeeded, it will only ring for that time!
I think you basically have to pick one or the other. Either set a long
timeout (15-30 sec, e.g. Dial(SIP/whatever,20) or don't use this feature.
The good news is that if the destination SIP server is actually
unreachable, Dial() should return almost instantly, at which point it
should jump to the failure priority.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
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--
---
Ricardo Carvalho
ITEC / IRICUP / Reitoria UP
tel: +351220408108
sip:[EMAIL PROTECTED]
[EMAIL PROTECTED]
---
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[asterisk-users] Forward to my phones the domain of the CALLERID in incoming URI calls
Is there a way I can forward to my phones the domain of the CALLERID in the CALLERID(number) field of INVITE messages, when some call arrives to my Asterisk? What happens in my architecture is this: INVITE [EMAIL PROTECTED] INVITE [EMAIL PROTECTED]'s_IP --- Asterisk --- john's_phone From: Mary sip:[EMAIL PROTECTED]From: Mary sip:[EMAIL PROTECTED]'s_IP As shown, Asterisk substitutes the domain of the caller contact in the From field of INVITE messages that are sent to the destination phone by Asterisk's IP address. That way, our phones just display Mary and mary when I want them to display Mary and [EMAIL PROTECTED], so that john can be aware that Mary is from an outside domain. Any ideas? How should be my extensions.conf so this can be possible? Regards, Ricardo. ___ --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] ENUMLOOKUP well succeeded but callee server unreached
I use ENUM lookup in my dialplan before sending calls through my PSTN trunk. One problem arises... When ENUMLOOKUP finds an SIP contact for that e164 number, Asterisk dials that contact, but when the remote server that should answer the call is down, or the IP link is down for some reason, the dial to PSTN trunk (which has the next priority) only takes place after the ring time of Dial application has expired (although it didn't even ring, but Asterisk keeps sending INVITEs to the callee server until Dial timeout). Is there any way I can force my dialplan to skip to the next priority in a few seconds (dialing my PSTN trunk) if there is no response from the callee server at the INVITEs sent my my server? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how can qualify=yes trigger some external event?
Hi all, The option qualify=yes allows Asterisk to check if it can reach the peer. If the device does not answer within the time-out period, Asterisk considers the device off-line for future calls. Is it possible to use this feature to trigger some external event, in case of failed reply from the peer that is tried to be reached? How can that be done? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP OPTIONS triggering some action in case of no reply
Hi, Is it possible to implement some kind of alarmist triggering some action, by sending SIP OPTIONS messages regularly to check that other peer is still online? I'm using Asterisk version 1.2.11 which I know it doesn't have any SNMP module, just 1.4 branch is being developing one; but is it any other way in Asterisk 1.2 to trigger some action in case of failure of SIP OPTIONS response? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Pickup with more than one argument
Dear all, I tried to use the following sintax to implement call pickup in Asterisk 1.2.17 with no success: exten = _**5219/5215,1,Pickup(5219) exten = _**5219/5215,2,Pickup(220408108) exten = _**5219/5215,3,Hangup Asterisk seems to just do the first priority command (Pickup(5219)) and if the ringing call comes from the channel 220408108, it doesn't jump to the second priority command. I've also tried to do it in only one line, like this: Pickup(5219220408108) but it doesn't work! After reading in voip-info, only in Asterisk 1.3 development, this issue has been considered to be implemented... I wonder if Asterisk 1.4 implements this since no version 1.3 has been released! Other option seems to be the use of Pickup2, but is it a stable option to implement in a production system? Thanks in advance, Ricardo. ___ --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] Call Pickup with more than one argument
Dear all, Does Pickup application accept multiple extensions pickup syntax, like the following line? Pickup(extension1extension2...) I've tried it in Asterisk 1.2.17 but it doesn't work. Does it work in Asterisk 1.4 already? Or is any other way in any version of Asterisk that I can use to do the same thing? Thanks, Ricardo. ___ --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] unconditionally redirecting incoming calls by 302 Moved Temporarily messages doing right accounting
Dear all, In my Asterisk 1.2.17 architecture different levels of permissions are established using different contexts that hierarchically include more permissive contexts until default context is reached. In default context there are only local extensions, only in more restricted contexts there are the PSTN access. So, if some user dials some number, Asterisk looks which context that user belongs to in sip.conf and sends that call to that context in extensions.conf. Call flow goes successively including other contexts along the hierarchy until some established filter matches, and than that call is routed to the destination. If no match is found after call flow has descend until the default context, Asterisk hungs up the call. Problem arises when The problem is that the phones I've deployed in my site have the optional feature of unconditionally redirecting incoming calls to other phone number by sending a 302 Moved Temporarily SIP message back to Asterisk, carrying the new contact that should be dialled by the server. When this happens, Asterisk seems to send this 302 message to the default context. If the new contact is some internal extension, it matches some rule in the default context, and Asterisk dials that extension with no problem. If the new contact is some PSTN number, Asterisk can't find a successful matching rule in default context because only upper hierarchy contexts match PSTN numbers, and call is hung up. To solve this, I can include PSTN numbers matching rules in default context (or include upper hierarchy permission contexts in default), but than, every one without PSTN dial permissions would be able to dial PSTN numbers!! Is there any way that I can make that 302 message be dropped in the context to which the user that redirected the call belongs to, and not the default context, because, this is the one that should be charged for the forwarded accounting? And like this, the redirected call would only take place if the user that redirected the call has PSTN permissions to do that! Thanks in advance, Ricardo. ___ --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] Only secretary can call the boss, all others only reach the secretary when dial the boss extension
With Ioan suggestion it still doesn't work, because Asterisk still
thinks that the INVITE sent as consequence of the REFER message isn't
correlated with a transferred call coming from the secretary.
I've also tried to do it using different contexts, but it still doesn't
work. I've done like this:
[default]
exten = secretary_extension,1,Dial(SIP/secretary_extension)
exten = boss_extension,1,Dial(SIP/secretary_extension)
[secretary]
include = default
exten = boss_extension,1,Dial(SIP/boss_extension)
The problem seems to be that in either case, Asterisk doesn't keep the
state of the call, to know that if transferred from the secretary, the
server should let it pass to the boss and not redirecting it back to the
secretary.
May this be solved with Transfer([Tech/]dest[|options])? And is it the
only way to do it? Can't it be done with normal transfer key that the
phones I've deployed have?
Any other ideas?!
Thanks,
Ricardo.
Ioan Indreias wrote:
Maybe you could use something like:
exten =
boss_ext,1,GotoIf($[${CALLERID(number)}=secretary_ext]?boss:secretary)
exten = boss_ext,n(boss),Dial(SIP/boss_ext)
exten = boss_ext,n(secretary),Dial(SIP/secretary_ext)
## nini @ www.modulo.ro ##
Jonathan k. Creasy wrote:
Why don't you just give the secretary the boss' REAL extension and
give a different extension to the world that just rings the secretary?
-jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho
Sent: Friday, January 26, 2007 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Only secretary can call the boss, all others
only reach the secretary when dial the boss extension
Dear all,
How may I configure my extensions.conf so that only the boss's
secretary
can call the boss through his extension, all others when dial his
extension only makes the boss's secretary phone ring, not his. If she
wants, she can transfer the incoming call to the boss dialling his
extension.
I've tried the following, but it doesn't work:
exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)
exten = _boss_extension,1,Dial(SIP/secretary_extension)
This doesn't work because when the secretary tries to transfer the call
to the boss (using her phone's transfer key, not #), one REFER SIP
message is sent back to the caller's phone providing him the new
address
for whom the next INVITE should be sent. That INVITE is sent, but when
reaches Asterisk, that INVITE matches this line:
exten = _boss_extension,1,Dial(SIP/secretary_extension)
and not this one:
exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension)
Any ideas of how may I solve this issue?
Regards,
Ricardo.
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11:11 AM
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[asterisk-users] How to match wild card inside a GoToIf?
How can I match wildcards inside a GoToIf?
I have something like this, but it doesn't work:
[default]
exten = _2,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,GotoIf($[${ARG1} = 220408XXX]?2:3)
exten = s,2,Hangup
Any ideas?
Regards,
Ricardo.
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[asterisk-users] BLF not working with Asterisk 1.4.0
Dear all, I've implemented BLF for use with some Grandstream GXP-2000 phones and it works fine in 1.2.x versions of Asterisk, although I tested it with version 1.4.0 and it doesn't work! Has the needed syntax changed for configure BLF for this version of Asterisk? It it a bug of this version? Or should it be misconfiguration that I'm doing? Thanks, Ricardo. ___ --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] FAX using T38
Only Asterisk 1.4.0 has chan_sip.c with T.38 code in it. Although it still doesn't work because it's full of bugs. Seems to me that developers just pasted the T.38 patch code from the branch developing that issue, and nothing else have done to improve it. It has to be debugged. Regards, Ricardo. Zoa wrote: So does asterisk (Albeit with a commercial package) http://www.attractel.com/t38.html Lee Howard wrote: Matt Riddell [NZ] wrote: Does OpenPBX do a T.38 gateway then? Yes, it does. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.0
Try this: /etc/init.d/zaptel start Than do lsmod |grep zaptel and it should show zaptel loaded Ricardo. Mike Hammett wrote: I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make install and I don’t see any errors. This is out of my modprobe.conf: install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp /sbin/ztcfg install ztdynamic /sbin/modprobe --ignore-install ztdynamic /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth /sbin/ztcfg install wct1xxp /sbin/modprobe --ignore-install wct1xxp /sbin/ztcfg install wcte11xp /sbin/modprobe --ignore-install wcte11xp /sbin/ztcfg install pciradio /sbin/modprobe --ignore-install pciradio /sbin/ztcfg install ztd-loc /sbin/modprobe --ignore-install ztd-loc /sbin/ztcfg install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg alias wcfxs wctdm alias wct2xxp wct4xxp install zttranscode /sbin/modprobe --ignore-install zttranscode /sbin/ztcfg install wct4xxp /sbin/modprobe --ignore-install wct4xxp /sbin/ztcfg However: [EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel FATAL: Module zaptel not found. /var/log/dmesg doesn’t say anything about zaptel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple phones registered for the same user
Dear all, I've noticed that when I have a phone registered in Asterisk, and then I register another phone with the same user, the sip show peers in the CLI shows that Asterisk replaced the IP of the first phone by the IP of the last one registered for that user. Consequently, if someone calls that user, only the last phone rings!! How may I configure Asterisk to be able to fork all incoming calls to every phones registered for each user, so that every phone ring until someone answers the call in one of them? Thanks in advance, Ricardo. ___ --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] multiple phones registered for the same user
Can't I register multiple phones with the same user/password? That's what I pretend to do, not ring groups... Thanks, Ricardo. Azfhasterisk wrote: Create a different user for each phone and create a ring group with the phones that you want to ring. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Wednesday, February 28, 2007 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] multiple phones registered for the same user Dear all, I've noticed that when I have a phone registered in Asterisk, and then I register another phone with the same user, the sip show peers in the CLI shows that Asterisk replaced the IP of the first phone by the IP of the last one registered for that user. Consequently, if someone calls that user, only the last phone rings!! How may I configure Asterisk to be able to fork all incoming calls to every phones registered for each user, so that every phone ring until someone answers the call in one of them? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple phones registered for the same user
Too bad... Thanks for all replays. Regards, Ricardo. Eric ManxPower Wieling wrote: Ricardo Carvalho wrote: Can't I register multiple phones with the same user/password? That's what I pretend to do, not ring groups... No, you cannot register multiple phones with the same user/password. ___ --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] Ex-Girlfriend syntax and RealTime Extensions
As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please give me one example. Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Radius users authentication
Thanks yusuf,
Any other experience on this subject? Anyone knows if Asterisk 1.4
already implement Radius authentication properly? Has anyone ever
patched Asterisk with the patch from the Digium Issue Tracker available
in the URL: http://bugs.digium.com/view.php?id=5424 and got well succeeded?
Thanks once again,
Ricardo.
yusuf wrote:
Ricardo Carvalho wrote:
Dear all,
I've searched the web about Asterisk with Radius integration for user
authentication, and got a bit confused...
I see that there have been some work around it, there is PortaOne's
Radius client patch, an still open branch of Digium Issue Tracker
SIP peer authentication on an external database (RADIUS - LDAP),
etc. Although, none of these seems to give me the confidence to
implement it in a production environment...
What do you people recommend me? Which Asterisk+Radius solution
should in your opinion be the best choice? Does Asterisk 1.4 already
implement it properly?
Thanks in advance,
Ricardo.
Here is a mock-up of what I used to hook-up to a Radius Server, with
Porta's patch. It worked quite well for us. I have'nt used it in 2
years or so, cant remember much :) . I thin we got it to work by
seeing the debug (set it in /etc/asterisk/logger.conf) and seeing what
values were getting sent and recieved.
;exten = _X.,1,SetVar(RADIUS_Server=x.x.x.x)
exten = _X.,2,SetVar(RADIUS_Secret=secret)
exten = _X.,3,SetVar(NAS_IP_Address=x.x.x.x)
exten = _X.,4,SetVar(CALLERID=${CALLERIDNUM})
exten = _X.,5,SetVar(DNID=${EXTEN})
;
; Set account to authorize by
; It can be a prepaid calling card PIN, ANI, or SIP ID depending on
your application
;
;exten = _X.,6,SetAccount(${CALLERIDNUM})
exten = _X.,6,SetAccount(${CALLERIDNAME})
;
; RADIUS Authorize
; Called as:
agi-rad-auth.pl|parametr1=value1parametr2=value2parametr3=value3
; Possible parametrs:
; Routing=XXX will will send h323-ivr-out = 'PortaBilling_Routing:XXX'
attribure (XXX is usually SIP)
; AuthorizeBy=SIP requires
SIPGetHeader(SIP_Authorization=Proxy-Authorization) first +
externalauth=yes in sip.conf
; AuthorizeBy=Account requires SetAccount(username) first
; Password=Password optional and may be used together with
AuthorizeBy=Account
; IfFailed=DoNotHangup optional, used for custome authentication error
processing i.e. IVR
;
;
exten =
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=AccountPassword=${CALLERIDNUM}IfFailed=DoNotHangup
;exten =
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=AccountPassword=AccountIfFailed=DoNotHangup
;exten =
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=AccountIfFailed=DoNotHangup
;
exten = _X.,8,NoOp(${h323-credit-time})
exten = _X.,9, Set(TIMEOUT(absolute)=${h323-credit-time:17})
;exten = _X.,10, AbsoluteTimeout(${h323-credit-time})
exten = _X.,10,Goto(sip-calls,${EXTEN},1)
exten = _X.,11,Hangup
exten = T,1,NoOp(timeout)
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[asterisk-users] Monitoring which users are online in realtime
Hi all, Is there a way to keep track in Asterisk of which phones are online in realtime using some MySQL DB table for exemple, much like sip show peers does in the CLI? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinct call permissions for each user
Thanks Luki, that's exactly what I was looking for, I'll give it a try...
Regards,
Ricardo.
Luki wrote:
someone please give me one example?
[locals]
exten = _NXX,1,Macro(outcall,${EXTEN})
[longdistance]
exten = _1NXXNXX,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,Dial(SIP/[EMAIL PROTECTED])
exten = s,2,Dial(Zap/.../${ARG1})
[fullaccess]
include = locals
include = longdistance
include = ...
[restricted]
include = locals
include = ...
Put user A into the restricted context, and user B into the fullaccess
context. You can include other extension (i.e. services) and implement
roll-over onto a backup trunks in macro-outcall.
You can of course also simply it and only have two contexts and no
macro, etc.
--Luki
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[asterisk-users] Asterisk with Radius users authentication
Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch, an still open branch of Digium Issue Tracker SIP peer authentication on an external database (RADIUS - LDAP), etc. Although, none of these seems to give me the confidence to implement it in a production environment... What do you people recommend me? Which Asterisk+Radius solution should in your opinion be the best choice? Does Asterisk 1.4 already implement it properly? Thanks in advance, Ricardo. ___ --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] Distinct call permissions for each user
Dear all, How may I configure my extensions.conf to stablish different PSTN access permissions for each user, letting for example user_A make only local calls and user_B make local and long-distance calls? I guess it can be done using include of other contexts, but how exactly? someone please give me one example? Thanks in advance, Regards, Ricardo. ___ --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] Only secretary can call the boss, all others only reach the secretary when dial the boss extension
Dear all, How may I configure my extensions.conf so that only the boss's secretary can call the boss through his extension, all others when dial his extension only makes the boss's secretary phone ring, not his. If she wants, she can transfer the incoming call to the boss dialling his extension. I've tried the following, but it doesn't work: exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension) exten = _boss_extension,1,Dial(SIP/secretary_extension) This doesn't work because when the secretary tries to transfer the call to the boss (using her phone's transfer key, not #), one REFER SIP message is sent back to the caller's phone providing him the new address for whom the next INVITE should be sent. That INVITE is sent, but when reaches Asterisk, that INVITE matches this line: exten = _boss_extension,1,Dial(SIP/secretary_extension) and not this one: exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension) Any ideas of how may I solve this issue? Regards, Ricardo. ___ --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] Call Transfers in SER + Asterisk architecture
Hi Marco, Ser has IP of Asterisk server in his trusted table, Asterisk isn't registered in Ser. When Ser needs to use Asterisk, it simply rewrites the IP destination with Asterisk's IP, and routes them to him. For example, here's one failed attempt in transferring a call PSTN - VoIP - VoIP: PSTN Asterisk Ser phone_A phone_B |INVITE| | | | | -- | | | | | 100 Trying | | | | | --- | | | | | | INVITE| | | | | -- |INVITE | | | | | --- | | | | |100 trying | | | 100 trying | --- | | | 100 trying | --- | 180 Ringing | | | -- | 180 Ringing | --- | | | 180 Ringing | -- | | | | -- | | | | | ACK | | | | | --- | ACK | | | | | --- | ACK | | | | | --- | | | | RTP | | | | == | | | | | | | | | | REFER | | | | REFER| --- | | | | -- | | | | | 404 Not Found | | | | | --- | 404 Not Found | | | | | -- | | | | | | | In this example, phone_A answers the PSTN originated call, and wants to transfer the call to phone_B. A REFER message is than routed backwards to Asterisk, and he replies with those 404 Not Found messages. Phone_B never gets called! Should Asterisk be registered in Ser so this can work properly? How can that be done? Thanks, Ricardo. Marco Mouta wrote: Hi Ricardo, Could you post a specific example where your problem happens. That way would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, *Ricardo Carvalho* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options
[asterisk-users] Call Transfers in SER + Asterisk architecture
Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to disable the 482 Loop Detected messages sent by Asterisk
Is there a way to make Asterisk don't send 482 Loop Detected error messages and continue with the transaction that is taking place? Thanks, Ricardo. ___ --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] problem with redirects
Dear all, My architecture is having some problems with redirects. In the following diagram is shown a simple erroneous test. When someone dials from the PSTN, signalling of the incoming call is passed to Asterisk which routes to SIP Express Route (Ser), and then Ser routes to the phone. The user has configured the phone to forward all calls to another PSTN number, and then, a 302 Moved Temporarily reply goes back to Ser which forwards back to Asterisk. Because Asterisk is configured with promiscredir=yes, it sends a reINVITE to the number announced in the 302 message as expected, and then that new INVITE goes back to Ser. Ser looks at the called number in that INVITE and because it is a PSTN number, sends the call back to Asterisk so this gateway can route it to PSTN. Because Asterisk receives the last INVITE with the same Call-ID that he passed to Ser in the anterior INVITE, he thinks it's a loop, and ends the communication with a 482 Loop Detected message. How can I configure Asterisk so that he can route the last INVITE to PSTN without giving me that error? PSTN Asterisk Ser UAC |INVITE| | | | -- | | | | 100 Trying | | | | --- | | | - | | INVITE| INVITE| | | -- | --- | | | 100 trying | 100 trying | | | --- | | | | 302 Moved Temporarily | 302 Moved Temporarily | | | -- | --- | | | ACK | ACK | | | --- | --- | - | | INVITE| | | | --- | | | | 100 trying | | | | --- | | | | INVITE| | | | --- | | | | 482 Loop Detected | | | | --- | | | | ACK | | | | --- | | Thanks in advance, Ricardo. ___ --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] problem with redirects
OK, to simplify the reading I'll resume my problem... Is there a way to make Asterisk send a call to Ser witch reroutes it back to the same asterisk server ,without resulting in a loop detected error in Asterisk? Thanks, Ricardo. ___ --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] FAX using T38
Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in Asterisk. Although I can't send FAXes to PSTN using T38. The problem, as far as I know, might be assigned with the Content-Length shown in the message header of every SIP message negotiating T38 parameters. I've observed that after leaving Asterisk, the Content-Length of every message carrying T38 parameters gets shorter than truly is, and contrarily to my ATAs that seem to don't care about this, my Telco analyses the packet length written in this messages and truncates them, aborting the call. Does anyone experienced this too? Any ideas besides editing the chan_sip.c code to fix this problem? Thanks, Ricardo. ___ --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] special characters in alphanumeric extensions
Hi all, I use alphanumeric names as extensions in my Asterisk architecture, which are the username part of the e-mail of each person at my site. Because Asterisk was primarily built to use numeric extensions, I'm having some problems with people that have usernames with dots between letters, like john.doe. More specifically my problem is when john.doe dials some number. Asterisk doesn't match his rule in extensions.conf. I have in that file the following line: exten = _[0-9]./john.doe,1,Dial(SIP/[EMAIL PROTECTED],60) When that user dials some number, Asterisk never matches his rule. This only happens because dots are special parameters for Asterisk. I've tried to put a slash \ before the dot, but nothing happens!... Any suggestion? Thanks in advance, Ricardo. ___ --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] Glitches in sound every time that Asteriskreceives reINVITEs
In fact as far as I know, Asterisk stands in the middle of calls, breaking one transaction and initiating another to the other side, doing the bridge between them... Although good in some cases like permitting to start a new transaction to the next hop changing codecs, in my case I don't need that feature because I'm using reINVITEs to implement session-timer support in the user agent to solve problems of whong accounting if power failure or link happens... Is there any way to disable those breaks in audio stream? Regards, Ricardo. Andreas Sikkema wrote: My Asterisk server is working fine, although every time that in the middle of any call there is a reinvite, the user hears a glitch. Why is this happening? How can I solve this problem? That's because a REINVITE is generally used to change from one codec to another. For some reason this involves stopping the existing audio, waiting a little while and then starting a new audio stream. So far this one of the reasons why I don't like reinvite... ___ --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] anti ex-girlfriend
Has far as I know, Asterisk doesn't support ex-girlfriend logic in realtime extensions so far. Regards, Ricardo. Pezhman Lali wrote: Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 |2 | hangup | | 455 | DID | 14193016880 |1 | Dial | H323/[EMAIL PROTECTED]|60 | didx.org for test by pezhman it's work good. but for routing by caller id: 456 | DID | 14193016880/2085838 |2 | hangup || 455 | DID | 14193016880/2085838 |1 | Dial | H323/[EMAIL PROTECTED]|60 | didx.org for test by pezhman this extension does not work , with a call from 2085838 please help me tanx Pezhman We have the perfect Group for you. Check out the handy changes to Yahoo! Groups (http://groups.yahoo.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail usernames can't begin with j letter?
Thanks for all that replayed, the problem is solved! Regards, Ricardo. ___ --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 usernames can't begin with j letter?
Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with j letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for 'ohn' (for a called user named john, for example) Is this some kind of Asterisk bug, or am I skipping some configuration? How can I make things work fine? Thanks in advance, Ricardo. ___ --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_HEADER function; what names are available?
Any news on this thread? I also need to know the way to get the R-URI
from sip INVITE messages received by Asterisk, through ${SIP_HEADER()}.
Thanks in advance,
Ricardo.
kjcsb wrote:
I have read the wiki about the SIP_HEADER function (http://www.voip-
info.org/wiki/index.php?page=Asterisk+func+sip_header). Where can I
get a list of the names that are available to be used with the
function e.g. TO is one name as in ${SIP_HEADER(TO)}. What are the
others?
I would guess that you can check the RFC. Easier is to turn on SIP
debug and see the INVITE packet yourself and
check the headers that you have with your equipment.
/Olle
Thanks but I don't know how to get the actual INVITE details (the
request URI?). For example I want to get sip:[EMAIL PROTECTED]
SIP/2.0 from the following dialogue:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:147.202.nn.nnn;ftag=bf7eced18eb7271b;lr=on
Via: SIP/2.0/UDP 147.202.nn.nnn;branch=z9hG4bKe49c.21b320a3.0
Via: SIP/2.0/UDP 60.234.nnn.nnn;branch=z9hG4bK76bf3dec8d45b972
From: User sip:[EMAIL PROTECTED];tag=bf7eced18eb7271b
To: sip:[EMAIL PROTECTED]
etc
I can get Record-Route, Via, From, To etc but don't know how to get
the bit after the INVITE. Interestingly only the first Via is returned
by ${SIP_HEADER(VIA)}.
I've tried R-URI, RURI, URI, ALL, *, blank.
Any advice appreciated.
Cameron
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Re: [asterisk-users] voicemail usernames can't begin with j letter?
I'm running Asterisk version 1.2.10. I also tried with version 1.2.4 and
got same problem.
I use SIP and in my extensions.conf I have the following code:
exten = _[a-z].,1,Answer
exten = _[a-z].,2,Wait(1)
exten = _[a-z].,3,VoiceMail(${EXTEN})
exten = _[a-z].,4,Hangup
Through my testing I found that the problem is that when someone enters
for example john's voicemail, Asterisk thinks that j letter is jump
flag to n+1 priority. How can I disable, (if possible) this erroneous
interpretation that Asterisk does?
Regards,
Ricardo.
Bruce Reeves wrote:
What version of * are you running? I have several j usernames in
voicemail.conf under SVN-branch-1.2-r37458M.
On 10/20/06, *Ricardo Carvalho* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Dear all,
I've configured Asterisk Voicemail, but after some tests I
realised that
when some call is sent to the voicemail of someone which username
begins
with j letter, Asterisk gives me the error:
WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in
voicemail config file for 'ohn'
(for a called user named john, for example)
Is this some kind of Asterisk bug, or am I skipping some
configuration?
How can I make things work fine?
Thanks in advance,
Ricardo.
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Bruce
Nortex Networks
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[asterisk-users] Does Asterisk 1.4 going to support realtime ex-girlfriend logic?
Hi all, I was deploying Realtime Extensions when I realised that Realtime Asterisk yet doesn't support ex-girlfriend logic, what made me abandon that implementation! Does Asterisk 1.4 going to support that feature? Regards, Ricardo. ___ --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: Can you explain why multiple registration is an important (missing) feature ?
You can use Asterisk along with Ser. Asterisk for advanced features like Voicemail and gateway, and Ser for routing SIP messages, Registrar, acc, etc. Take a look at: http://www.voip-info.org/wiki-Asterisk+at+large It works!! Regards, Ricardo. Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In some cases : Yes. But we have the following situation : We re using cisco 7960 phones in each office (about 150 of them) Do Cisco phones support paging/intercom? If yes, please send me link to some useful pages. Now we want to give the user's the ability to take their number with them. So when you change places you can call a defined number which will write you a config file for your new phone. To much work. Is it working right? Now, if I have extension 1234 and go to a different office, or to a meeting room, etc and log into that phone using my extension, if i did not log out my normal phone we have a problem because we have to SIP/1234. I haven't found a good solution for that yet, but if I could register two SIP/1234 phones the problem would be solved. I would like that Asterisk supports multiple registers, but till then you could use dynamic agents. Agent can log in from every phone. And you send incoming phone call to agent instead to extension. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail adjustments
Hi all,
Some questions about Asterisk Voicemail adjustments I want to make:
- how can I limit the number of voicemail messages stored per user in
their voicemail folder?
(to expire voicemail after a specified number of days I know that there
is in /contrib/scripts one script to do that)
- how can I turn the voicemail messages built according to the syntax in
voicemail.conf file, to show the ${VM_DATE} parameter in other languages?
- how can I substitute the vm-intro, auth-thankyou and vm-goodbye
recordings to recordings in other languages?
Thanks,
Ricardo.
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[asterisk-users] Thomson 2030
Hi all, Does Thomson 2030 hardphone has the feature of supporting more than one user registered at the same time? I heard not... But I think that's weird because it has 4 profiles... Thanks, Ricardo. ___ --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] Grandstream Budgetone phones don't show
Thanks Jessee, I've just sent an e-mail to Grandstream support asking if they are planning in a near future to release a firmware implementing alphanumeric callerid for Budgetone series. When they answer me, I'll replay to this thread with their feedback, so the community can also benefit... Regards, Ricardo. Jessee J Holmes wrote: Ricardo, From what I know its a physical limitation of the display Grandstream chose on that phone, Grandstream recommends purchasing the GXP-2000 phone instead if you're looking for this feature. Grandstream has no plans from what I am aware of of making this change to the BudgetTone series phones. You are more than welcome to inquire directly from Grandstream though, this is just from what I know from dealing with them in the past. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 11, 2006, at 11:47 AM, Ricardo Carvalho wrote: I guess this functionality will be in the future added to new firmware releases don't you people think so? Ricardo. Doug Lytle wrote: These phones aren't capable of alphanumeric entries, only numeric. Doug Tom Vile wrote: They only do numeric callerid. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right
Grandstream support just answered me saying that: BT100/200 LCD does not supports alphanumeric caller ID display. You may want to try GXP-2000.. It's confirmed! Future firmwares won't support that feature! :( Thanks to all that replied, Regards, Ricardo. Craig Guy wrote: The lcd in the current budgetone series cannot support alphnumeric display. Craig - Original Message - From: Ricardo Carvalho [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 12, 2006 8:11 PM Subject: Re: [asterisk-users] Grandstream Budgetone phones don't show Thanks Jessee, I've just sent an e-mail to Grandstream support asking if they are planning in a near future to release a firmware implementing alphanumeric callerid for Budgetone series. When they answer me, I'll replay to this thread with their feedback, so the community can also benefit... Regards, Ricardo. Jessee J Holmes wrote: Ricardo, From what I know its a physical limitation of the display Grandstream chose on that phone, Grandstream recommends purchasing the GXP-2000 phone instead if you're looking for this feature. Grandstream has no plans from what I am aware of of making this change to the BudgetTone series phones. You are more than welcome to inquire directly from Grandstream though, this is just from what I know from dealing with them in the past. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 11, 2006, at 11:47 AM, Ricardo Carvalho wrote: I guess this functionality will be in the future added to new firmware releases don't you people think so? Ricardo. Doug Lytle wrote: These phones aren't capable of alphanumeric entries, only numeric. Doug Tom Vile wrote: They only do numeric callerid. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can Asterisk bind on multiple ports?
Can Asterisk bind on multiple ports? I wish I could in my sip.conf make Asterisk bind different ports per different context, so that calls coming in udp port 5060 would fall in one context and calls coming in port 5061 fall in other different context. Is that possible? How can I edit my sip.conf to be able to do that? Regards, Ricardo. ___ --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] Grandstream Budgetone phones don't show alphanumeric caller right
I have tested Grandstream Budgetone 102 and Grandstream Budgetone 200 and with both, if they are called from a caller that is an alphanumeric user, their display shows a unintelligible name impossible to figure out who is calling!! If the caller is a numeric one, in both phones their display shows correctly the caller's contact. I've updated their firmwares to the latest ones and that problem persists... Does anybody also experienced this? Thanks, Ricardo. ___ --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] Grandstream Budgetone phones don't show alphanumeric caller right
Thanks Tom, That's too bad... right now that I was thinking about buying them to a mass deployment environment... Regards, Ricardo. Tom Vile wrote: They only do numeric callerid. On 9/11/06, *Ricardo Carvalho* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have tested Grandstream Budgetone 102 and Grandstream Budgetone 200 and with both, if they are called from a caller that is an alphanumeric user, their display shows a unintelligible name impossible to figure out who is calling!! If the caller is a numeric one, in both phones their display shows correctly the caller's contact. I've updated their firmwares to the latest ones and that problem persists... Does anybody also experienced this? Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] Grandstream Budgetone phones don't show
I guess this functionality will be in the future added to new firmware releases don't you people think so? Ricardo. Doug Lytle wrote: These phones aren't capable of alphanumeric entries, only numeric. Doug Tom Vile wrote: They only do numeric callerid. ___ --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] distinguishing users by their domain
In extensions.conf I want to implement a dial plan that distinguishes the users that wish to dial a PSTN number by their own domain, so that [EMAIL PROTECTED] goes out to PSTN by a different DID than [EMAIL PROTECTED] I tried the following line, but that doesn't distinguish between domains, and then if [EMAIL PROTECTED] or [EMAIL PROTECTED] dials some PSTN number, both calls goes out using same DID (did1): exten = _[0-9]./john,1,Dial(SIP/[EMAIL PROTECTED],120) I tried then using the following lines: exten = _[0-9]./[EMAIL PROTECTED],1,Dial(SIP/[EMAIL PROTECTED],120) exten = _[0-9]./[EMAIL PROTECTED],1,Dial(SIP/[EMAIL PROTECTED],120) But those syntax doesn't work. How can I do it? Any clues? Thanks, Ricardo. ___ --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] distinguishing users by their domain
So... does anybody know how can I do this? Maybe using a way to distinguish users not by their username, but by other fields of SIP INVITE messages? Regards, Ricardo. Ricardo Carvalho wrote: In extensions.conf I want to implement a dial plan that distinguishes the users that wish to dial a PSTN number by their own domain, so that [EMAIL PROTECTED] goes out to PSTN by a different DID than [EMAIL PROTECTED] I tried the following line, but that doesn't distinguish between domains, and then if [EMAIL PROTECTED] or [EMAIL PROTECTED] dials some PSTN number, both calls goes out using same DID (did1): exten = _[0-9]./john,1,Dial(SIP/[EMAIL PROTECTED],120) I tried then using the following lines: exten = _[0-9]./[EMAIL PROTECTED],1,Dial(SIP/[EMAIL PROTECTED],120) exten = _[0-9]./[EMAIL PROTECTED],1,Dial(SIP/[EMAIL PROTECTED],120) But those syntax doesn't work. How can I do it? Any clues? Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk t.38 fax failed
In sip.conf add to [general] context and to every peer context that you want to register in Asterisk to use T.38 the following lines: t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no In udptl.conf file I have the following configurations: [general] udptlstart=4000 udptlend=4999 T38FaxUdpEC = t38UDPRedundancy T38FaxMaxDatagram = 400 udptlfecentries = 3 udptlfecspan = 3 Good luck, Ricardo. Kokfoo Soo wrote: Ricardo, Thanks, could you please share some of your t.38 passthrough configuration in sip.conf and also udptl.conf? Thanks, */Ricardo Carvalho [EMAIL PROTECTED]/* wrote: No, T.38 doesn't work with Asterisk. Only works with Asterisk t38passthrough patch that you can find at URL: http://bugs.digium.com/file_download.php?file_id=9335type=bug For me it only worked well with patch for version 1.2.4 of Asterisk. Regards, Ricardo. Kokfoo Soo wrote: Is T.38 fax work through Asterisk? I have the config below in my sip.conf, but the fax doesn't work and give me the CLI lines below. My current version is 1.2.10. Please help. [Inboundtopbx] type=friend context=pbx host=10.18.188.84 insecure=port dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 allow=ulaw t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no [OutboundfromPBX] type=peer host=10.18.161.222 canreinvite=no dtmfmode=rfc2833 disallow=all allow=g729 qualify=yes t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no -- SIP read from 10.18.188.84:50096: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.18.188.84:5060 From: ;tag=19D429E8-2084 To: ;tag=as3c87a22e Date: Tue, 05 Sep 2006 19:42:28 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK --- (9 headers 0 lines)--- Sep 5 15:30:31 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown SDP media type in offer: image 16406 udptl t38 Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. ___ --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 How low will we go? Check out Yahoo! Messenger’s low PC-to-Phone call rates. http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk t.38 fax failed
No, T.38 doesn't work with Asterisk. Only works with Asterisk t38passthrough patch that you can find at URL: http://bugs.digium.com/file_download.php?file_id=9335type=bug For me it only worked well with patch for version 1.2.4 of Asterisk. Regards, Ricardo. Kokfoo Soo wrote: Is T.38 fax work through Asterisk? I have the config below in my sip.conf, but the fax doesn't work and give me the CLI lines below. My current version is 1.2.10. Please help. [Inboundtopbx] type=friend context=pbx host=10.18.188.84 insecure=port dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 allow=ulaw t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no [OutboundfromPBX] type=peer host=10.18.161.222 canreinvite=no dtmfmode=rfc2833 disallow=all allow=g729 qualify=yes t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no -- SIP read from 10.18.188.84:50096: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.18.188.84:5060 From: sip:[EMAIL PROTECTED];tag=19D429E8-2084 To: sip:[EMAIL PROTECTED];tag=as3c87a22e Date: Tue, 05 Sep 2006 19:42:28 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK --- (9 headers 0 lines)--- Sep 5 15:30:31 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown SDP media type in offer: image 16406 udptl t38 Yahoo! Messenger with Voice. Make PC-to-Phone Calls http://us.rd.yahoo.com/mail_us/taglines/postman1/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com to the US (and 30+ countries) for 2¢/min or less. ___ --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
