[Asterisk-Users] Asterisk Newbie
Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is noa forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. That is why I will post my questions here: 1- Transcoding: is this when you go from g711 to g729 for example? Or when you go fromSIP to IAx? 2- What is the best GUI tool to configure * ? 3- Do I need to install a PCI (fxo or fxs) to have meetme, music onnhold etc? 4- IfI have a SIP device behind a firewall the supports SIP transformations (sonicwall pro230) and the * is outside the firewall, do I have to open ports 5060 anyway? What about the audio? Regards Fabian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hardware question
Hello I want to to know if the motherboards VIA are fully supporte by asterisk. And also, some of those motherboars say that with 1 pci slot , using a special riser card you can connect 2 pci cards. Will that work to have 2 pci cards (FXS or FXO ) on asterisk? thank you Fabian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T.38 passthru on 1.8.5
Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway -- asterisk -- Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of inactivity the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] same sip peer as user and provider
Hello Up to version 1.6.0 we have been able to configure the same SIP device as a user [inbound trunk] and as a peer [outbound trunk] w/o issues. After we switched to version 1.8 this setup wont work, apparently one can not have the same IP on 2 different trunks anymore. The trunk that is configured as user or friend is not choosen when the inbound call hits asterisk, instead the outbound trunk is and that trunk is usually w/o context and then asterisk can not find any call logic in the dialplan in the default extension, hence the call fails. as a workaround I have been trying the SIP_CODEC variables [inbound and outbound] but it wont help me in all cases. Also I can not set the ptime on the fly using those variables in the dialplan. After reading in the forums and the books/guides apparently the users are matched by the From header and peers are matched by IP. Is this is the intended behavior now? any help is greatly appreciated, txs a lot in advance fborot -- _ -- 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] same sip peer as user and provider
yes, same thing From: fbo...@hotmail.com To: fbo...@hotmail.com Subject: RE: same sip peer as user and provider Date: Tue, 30 Aug 2011 10:35:01 -0400 yes my friend. same thing From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: same sip peer as user and provider Date: Tue, 30 Aug 2011 10:16:11 -0400 Hello Up to version 1.6.0 we have been able to configure the same SIP device as a user [inbound trunk] and as a peer [outbound trunk] w/o issues. After we switched to version 1.8 this setup wont work, apparently one can not have the same IP on 2 different trunks anymore. The trunk that is configured as user or friend is not choosen when the inbound call hits asterisk, instead the outbound trunk is and that trunk is usually w/o context and then asterisk can not find any call logic in the dialplan in the default extension, hence the call fails. as a workaround I have been trying the SIP_CODEC variables [inbound and outbound] but it wont help me in all cases. Also I can not set the ptime on the fly using those variables in the dialplan. After reading in the forums and the books/guides apparently the users are matched by the From header and peers are matched by IP. Is this is the intended behavior now? any help is greatly appreciated, txs a lot in advance fborot -- _ -- 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] T.38 passthru on 1.8.5
both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway -- asterisk -- Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of inactivity the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] T.38 passthru on 1.8.5
will installing spandsp help with t.38 pass-through? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway -- asterisk -- Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of inactivity the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] T.38 passthru on 1.8.5
txs a lot for your explanation steve so, it should work w/o spandsp fairly fine if we do not have a bad connection. I see that this version has a lot of fixes related to t.38 but is the implementation already mature enough to guarantee a decent success rate with fax calls? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 13:15:19 -0400 will installing spandsp help with t.38 pass-through? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway -- asterisk -- Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of inactivity the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] T.38 passthru on 1.8.5
Txs a lot Kevin. I had just created and account on https://issues.asterisk.org/jira Let me know if this is the right place to post both the pcap capture and the sip logs. If not please help me out creating the account in the right place so that I can provide all the information you need. The sip debug logs I can post here but I need to change the real IPs, which is easy to do because it will be a text file. I appreciate your time and effort in helping us find the roout cause. Fborot From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 13:53:25 -0400 txs a lot for your explanation steve so, it should work w/o spandsp fairly fine if we do not have a bad connection. I see that this version has a lot of fixes related to t.38 but is the implementation already mature enough to guarantee a decent success rate with fax calls? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 13:15:19 -0400 will installing spandsp help with t.38 pass-through? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway -- asterisk -- Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of inactivity the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- 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] T.38 passthru on 1.8.5
Hi Kevin, I created the issue on the https://issues.asterisk.org/jira web site, posted the description of the prob and submitted asterisk console logs [sip and udptl debug on] and a wireshark capture taken on the asterisk machine showing both legs with signaling and media. PLease let me know what other thing you need you need me to provide. Again we thank you in advance fborot From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 19:24:15 -0400 Txs a lot Kevin. I had just created and account on https://issues.asterisk.org/jira Let me know if this is the right place to post both the pcap capture and the sip logs. If not please help me out creating the account in the right place so that I can provide all the information you need. The sip debug logs I can post here but I need to change the real IPs, which is easy to do because it will be a text file. I appreciate your time and effort in helping us find the roout cause. Fborot From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 13:53:25 -0400 txs a lot for your explanation steve so, it should work w/o spandsp fairly fine if we do not have a bad connection. I see that this version has a lot of fixes related to t.38 but is the implementation already mature enough to guarantee a decent success rate with fax calls? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 13:15:19 -0400 will installing spandsp help with t.38 pass-through? From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway -- asterisk -- Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of inactivity the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot
[asterisk-users] Asterisk removes SDP from 180 with SDP
Asterisk receives a 180 Ringing with SDP from the called side, then it sends 180 without SDP to the calling side. We would like asterisk to sends to the calling side the same response that was received from the called side. This is Asterisk cert 13.1, is that a new behavior, is there a setting to change this ? -- _ -- 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] Reply to INVITE with 1 codec
In Version 1.8 asterisk introduced this parameter preferred_codec_only, when set to yes the 200 OK to the INVITE contains 1 codec only from the available ones in the user sip profile. But in version 13.1 (I think version 11.2 also) is not working like that , it keeps sending all the codecs and sometimes both parties pick a different one causing one way audio. Example: INVITE has ulaw, alaw, gsm and 200 OK from asterisk has alaw, g729,ulaw. Then a media capture shows the calling side sending ulaw and the asterisk sends alaw causing one way audio. Is this happening to anybody else? This is the description of the parameter from the sip.conf preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer. -- _ -- 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] Question about Warning message
Starting with Asterisk 13.1 we are seeing this WARNING
messages a lot in our logs and console:
WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type
frames with SIP write)
We found that line in function sip_write inside chan_sip.c.
In our previous version (11.2.1) we did not see those messages being printed
(same verbosity level). We compared both versions of the functions and see no
difference at all in the 'default' switch case that handles that. We
think/assume that that function is being called in
different places on each version (11.2-1 vs 13-1).
We also think it has to do with the asterisk receiving rtp packets with comfort
noise which is not supported by asterisk.
We would like to know what can we do about it to behave more like the version
11?
We are not sure but could it be that version 11 handles it better ?. I am
attaching the functions on both versions for your review.
Thank you
/*! \brief Send frame to media channel (rtp) */
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct sip_pvt *p = ast_channel_tech_pvt(ast);
int res = 0;
switch (frame-frametype) {
case AST_FRAME_VOICE:
if
(!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast),
frame-subclass.format))) {
char s1[512];
ast_log(LOG_WARNING, Asked to transmit frame type %s,
while native formats is %s read/write = %s/%s\n,
ast_getformatname(frame-subclass.format),
ast_getformatname_multiple(s1, sizeof(s1),
ast_channel_nativeformats(ast)),
ast_getformatname(ast_channel_readformat(ast)),
ast_getformatname(ast_channel_writeformat(ast)));
return 0;
}
if (p) {
sip_pvt_lock(p);
if (p-t38.state == T38_ENABLED) {
/* drop frame, can't sent VOICE frames while in
T.38 mode */
sip_pvt_unlock(p);
break;
} else if (p-rtp) {
/* If channel is not up, activate early media
session */
if ((ast_channel_state(ast) != AST_STATE_UP)
!ast_test_flag(p-flags[0],
SIP_PROGRESS_SENT)
!ast_test_flag(p-flags[0], SIP_OUTGOING))
{
ast_rtp_instance_update_source(p-rtp);
if (!global_prematuremediafilter) {
p-invitestate =
INV_EARLY_MEDIA;
transmit_provisional_response(p, 183 Session Progress, p-initreq, TRUE);
ast_set_flag(p-flags[0],
SIP_PROGRESS_SENT);
}
}
p-lastrtptx = time(NULL);
res = ast_rtp_instance_write(p-rtp, frame);
}
sip_pvt_unlock(p);
}
break;
case AST_FRAME_VIDEO:
if (p) {
sip_pvt_lock(p);
if (p-vrtp) {
/* Activate video early media */
if ((ast_channel_state(ast) != AST_STATE_UP)
!ast_test_flag(p-flags[0],
SIP_PROGRESS_SENT)
!ast_test_flag(p-flags[0], SIP_OUTGOING))
{
p-invitestate = INV_EARLY_MEDIA;
transmit_provisional_response(p, 183
Session Progress, p-initreq, TRUE);
ast_set_flag(p-flags[0],
SIP_PROGRESS_SENT);
}
p-lastrtptx = time(NULL);
res = ast_rtp_instance_write(p-vrtp, frame);
}
sip_pvt_unlock(p);
}
break;
case AST_FRAME_TEXT:
if (p) {
sip_pvt_lock(p);
if (p-red) {
ast_rtp_red_buffer(p-trtp, frame);
} else {
if (p-trtp) {
/* Activate text early media */
if ((ast_channel_state(ast) !=
AST_STATE_UP)
!ast_test_flag(p-flags[0],
SIP_PROGRESS_SENT)
Re: [asterisk-users] Question about Warning message
thank you, we are using the same configuration files in 13, same setup, just different asterisk version. we just dont see the msgs in the console/logs, it is the same exact voice traffic on both asterisk versions is that something that you set on/off? if that is the case how can it be done? what is the alternative? what are their differences/characteristics? how to choose one over among others? thank you again From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: Question about Warning message Date: Mon, 23 Feb 2015 12:27:05 -0500 Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console: WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write) We found that line in function sip_write inside chan_sip.c. In our previous version (11.2.1) we did not see those messages being printed (same verbosity level). We compared both versions of the functions and see no difference at all in the 'default' switch case that handles that. We think/assume that that function is being called in different places on each version (11.2-1 vs 13-1). We also think it has to do with the asterisk receiving rtp packets with comfort noise which is not supported by asterisk. We would like to know what can we do about it to behave more like the version 11? We are not sure but could it be that version 11 handles it better ?. I am attaching the functions on both versions for your review. Thank you -- _ -- 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] set codec based on B side
Using Asterisk 18.12.0, a little confused on how to configure my pjsip.conf file to determine the codec to use for a call I have 2 endpoints: [Alice] disallow:all allow:ulaw,alaw,g729 [Bob] disallow:all allow:ulaw,alaw,g729 Alice calls into Asterisk on ext 100 and then we dial Bob I want to wait until Bod side codec is chosen to answer Alice and have each channel use the codec chose on Bob side. I see these options on this link, https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_pjsip codec_prefs_incoming_offer codec_prefs_outgoing_offer codec_prefs_incoming_answer codec_prefs_outgoing_answer but I dont see them on my pjsip.conf file. I only see these tow: incoming_call_offer_pref outgoing_call_offer_pref Do I have to use the 4 of them on each endoint Alice and Bob? Or just one side should be enough? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
