Hi guys, I'm trying to set up a SER server between 2 asterisk machines. I run into 2 issues. Whenever I call someone I don't get any ringback tone even so the call initiating asterisk machine gets the 180 message after 100. <--- SIP read from 10.4.1.80:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" sip:1000@82.98.89.134;tag=as4c964973 To: sip:017683035400@10.4.1.80 Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 out_uri=sip:017683035400@192.168.13.102:5060 via_cnt==1"
<-------------> --- (9 headers 0 lines) --- mg03*CLI> <--- SIP read from 10.4.1.80:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" sip:1000@82.98.89.134;tag=as4c964973 To: sip:017683035400@10.4.1.80;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2 Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 out_uri=sip:017683035400@192.168.13.102:5060 via_cnt==1
On SER I've configured to send this message: if (method=="INVITE") { if (uri =~ "sip:0[0-9]@*") { route(3); sl_send_reply("180", "Ringing"); break; } };
The other issue is that I don't see the caller id on the receiver side. I don't know if it is a asterisk or a SER issue. Only if I set the caller id on asterisk manual in extensions.conf with exten => _X.,1,Set(CALLERID(num)=06965006100) I'll see the caller id on the receiver side.
I would really appreciate any help Thanks Thorsten
Most probably your SER instance does not modify the callerid info so I would check both your asterisk configs and the configuration of your UAs. The ringback tone also looks like a configuration issue of your asterisk.
I would recommend you to get some info about the asterisk configuration to know which the problem might be.
Sam.
2008/4/29, Thorsten serusers@thorko.de:
Hi guys, I'm trying to set up a SER server between 2 asterisk machines. I run into 2 issues. Whenever I call someone I don't get any ringback tone even so the call initiating asterisk machine gets the 180 message after 100. <--- SIP read from 10.4.1.80:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134 sip%3A1000@82.98.89.134
;tag=as4c964973
To: <sip:017683035400@10.4.1.80 sip%3A017683035400@10.4.1.80> Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 sip%3A017683035400@10.4.1.80 out_uri=sip:017683035400@192.168.13.102:5060 via_cnt==1"
<-------------> --- (9 headers 0 lines) --- mg03*CLI> <--- SIP read from 10.4.1.80:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134 sip%3A1000@82.98.89.134
;tag=as4c964973
To: <sip:017683035400@10.4.1.80 sip%3A017683035400@10.4.1.80
;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2
Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 sip%3A017683035400@10.4.1.80 out_uri=sip:017683035400@192.168.13.102:5060 via_cnt==1
On SER I've configured to send this message: if (method=="INVITE") { if (uri =~ "sip:0[0-9]@*") { route(3); sl_send_reply("180", "Ringing"); break; } };
The other issue is that I don't see the caller id on the receiver side. I don't know if it is a asterisk or a SER issue. Only if I set the caller id on asterisk manual in extensions.conf with exten => _X.,1,Set(CALLERID(num)=06965006100) I'll see the caller id on the receiver side.
I would really appreciate any help Thanks Thorsten
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi samuel, I've already figured out the caller id issue. It was a mis configured asterisk. I set the caller id 1000 which isn't a international format. When I set it to callerid="Thorsten" <6965006100>" in sip.conf it works just fine. But I still have that ringback tone issue. Here comes the asterisk configuration. I've created an account on asterisk to connect my Snom phone to it [1000] type=friend username=1000 secret=mypass regexten=1000 host=dynamic context=toser callerid="Thorsten" <6965006577> qualify=yes nat=yes
The context in extensions.conf looks like this [toser] exten => _X.,1,Dial(sip/${EXTEN}@10.4.1.80)
I've also set the "progressinband" to "yes". When I make a call between the asterisk machines not going through SER it works. So I guess it is a SER issue. In the SER logs I don't see where it sends a proper 180 message, but I see it on the asterisk machine. So I don't know if "sl_send_reply" works. Thanks Thorsten
samuel wrote:
Most probably your SER instance does not modify the callerid info so I would check both your asterisk configs and the configuration of your UAs. The ringback tone also looks like a configuration issue of your asterisk.
I would recommend you to get some info about the asterisk configuration to know which the problem might be.
Sam.
2008/4/29, Thorsten <serusers@thorko.de mailto:serusers@thorko.de>:
Hi guys, I'm trying to set up a SER server between 2 asterisk machines. I run into 2 issues. Whenever I call someone I don't get any ringback tone even so the call initiating asterisk machine gets the 180 message after 100. <--- SIP read from 10.4.1.80:5060 <http://10.4.1.80:5060> ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134 <mailto:sip%3A1000@82.98.89.134>>;tag=as4c964973 To: <sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80>> Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 <mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134> CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 <http://10.4.1.80:5060> "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 <http://82.98.89.134> req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80> out_uri=sip:017683035400@192.168.13.102:5060 <http://sip:017683035400@192.168.13.102:5060> via_cnt==1" <-------------> --- (9 headers 0 lines) --- mg03*CLI> <--- SIP read from 10.4.1.80:5060 <http://10.4.1.80:5060> ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134 <mailto:sip%3A1000@82.98.89.134>>;tag=as4c964973 To: <sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80>>;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2 Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 <mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134> CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 <http://10.4.1.80:5060> "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 <http://82.98.89.134> req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80> out_uri=sip:017683035400@192.168.13.102:5060 <http://sip:017683035400@192.168.13.102:5060> via_cnt==1 On SER I've configured to send this message: if (method=="INVITE") { if (uri =~ "sip:0[0-9]@*") { route(3); sl_send_reply("180", "Ringing"); break; } }; The other issue is that I don't see the caller id on the receiver side. I don't know if it is a asterisk or a SER issue. Only if I set the caller id on asterisk manual in extensions.conf with exten => _X.,1,Set(CALLERID(num)=06965006100) I'll see the caller id on the receiver side. I would really appreciate any help Thanks Thorsten _______________________________________________ Serusers mailing list Serusers@lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers
Sorry guys, I missed one important point. The call initiating asterisk machine is using a public IP and the phone is in a private subnet as well as the SER server. So the entire constellation is like this
phone------------->asterisk--------->SER------->asterisk---------------->phone
192.168.9.14->82.98.89.134->10.4.1.80->192.168.13.102->192.168.9.15
I know it is completely weird, but this is only a test case, it isn't supposed to look like this in the final state. The other way around when coming from asterisk with a private IP it works. I tested this already. Do you know what options I've to set to send the ringback tone from SER to asterisk which has the public IP? Thanks Thorsten
Thorsten wrote:
Hi samuel, I've already figured out the caller id issue. It was a mis configured asterisk. I set the caller id 1000 which isn't a international format. When I set it to callerid="Thorsten" <6965006100>" in sip.conf it works just fine. But I still have that ringback tone issue. Here comes the asterisk configuration. I've created an account on asterisk to connect my Snom phone to it [1000] type=friend username=1000 secret=mypass regexten=1000 host=dynamic context=toser callerid="Thorsten" <6965006577> qualify=yes nat=yes
The context in extensions.conf looks like this [toser] exten => _X.,1,Dial(sip/${EXTEN}@10.4.1.80)
I've also set the "progressinband" to "yes". When I make a call between the asterisk machines not going through SER it works. So I guess it is a SER issue. In the SER logs I don't see where it sends a proper 180 message, but I see it on the asterisk machine. So I don't know if "sl_send_reply" works. Thanks Thorsten
samuel wrote:
Most probably your SER instance does not modify the callerid info so I would check both your asterisk configs and the configuration of your UAs. The ringback tone also looks like a configuration issue of your asterisk.
I would recommend you to get some info about the asterisk configuration to know which the problem might be.
Sam.
2008/4/29, Thorsten <serusers@thorko.de mailto:serusers@thorko.de>:
Hi guys, I'm trying to set up a SER server between 2 asterisk machines. I run into 2 issues. Whenever I call someone I don't get any ringback tone even so the call initiating asterisk machine gets the 180 message after 100. <--- SIP read from 10.4.1.80:5060 <http://10.4.1.80:5060> ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134 <mailto:sip%3A1000@82.98.89.134>>;tag=as4c964973 To: <sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80>> Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 <mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134> CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 <http://10.4.1.80:5060> "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 <http://82.98.89.134> req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80> out_uri=sip:017683035400@192.168.13.102:5060 <http://sip:017683035400@192.168.13.102:5060> via_cnt==1" <-------------> --- (9 headers 0 lines) --- mg03*CLI> <--- SIP read from 10.4.1.80:5060 <http://10.4.1.80:5060> ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134 <mailto:sip%3A1000@82.98.89.134>>;tag=as4c964973 To: <sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80>>;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2 Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 <mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134> CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 <http://10.4.1.80:5060> "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 <http://82.98.89.134> req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80> out_uri=sip:017683035400@192.168.13.102:5060 <http://sip:017683035400@192.168.13.102:5060> via_cnt==1 On SER I've configured to send this message: if (method=="INVITE") { if (uri =~ "sip:0[0-9]@*") { route(3); sl_send_reply("180", "Ringing"); break; } }; The other issue is that I don't see the caller id on the receiver side. I don't know if it is a asterisk or a SER issue. Only if I set the caller id on asterisk manual in extensions.conf with exten => _X.,1,Set(CALLERID(num)=06965006100) I'll see the caller id on the receiver side. I would really appreciate any help Thanks Thorsten _______________________________________________ Serusers mailing list Serusers@lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi Thorsten, I have seen your sip trace and I couldn't find SDP information in the 180 messages that you pasted. If you can check the " Content-Length: 0" it indicates if the SDP has or not information about media capabilities.
If I understand your test environment I supposed you have subscribers behind the Asterisk boxes, so the CPE subscriber should have the capability to generate a ringback inbound or outbound depends on if it receives or not a 180/183 message with SDP information (early media).
Regards
Alberto Cruz
-----Original Message----- From: serusers-bounces@lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Thorsten Sent: Tuesday, April 29, 2008 4:58 AM To: serusers@thorko.de Cc: serusers@lists.iptel.org Subject: Re: [Serusers] Ringback tone on SER
Sorry guys, I missed one important point. The call initiating asterisk machine is using a public IP and the phone is in a private subnet as well as the SER server. So the entire constellation is like this
phone------------->asterisk--------->SER------->asterisk---------------->pho ne
192.168.9.14->82.98.89.134->10.4.1.80->192.168.13.102->192.168.9.15
I know it is completely weird, but this is only a test case, it isn't supposed to look like this in the final state. The other way around when coming from asterisk with a private IP it works. I tested this already. Do you know what options I've to set to send the ringback tone from SER to asterisk which has the public IP? Thanks Thorsten
Thorsten wrote:
Hi samuel, I've already figured out the caller id issue. It was a mis configured asterisk. I set the caller id 1000 which isn't a international format. When I set it to callerid="Thorsten" <6965006100>" in sip.conf it works just fine. But I still have that ringback tone issue. Here comes the asterisk configuration. I've created an account on asterisk to connect my Snom phone to it [1000] type=friend username=1000 secret=mypass regexten=1000 host=dynamic context=toser callerid="Thorsten" <6965006577> qualify=yes nat=yes
The context in extensions.conf looks like this [toser] exten => _X.,1,Dial(sip/${EXTEN}@10.4.1.80)
I've also set the "progressinband" to "yes". When I make a call between the asterisk machines not going through SER it works. So I guess it is a SER issue. In the SER logs I don't see where it sends a proper 180 message, but I see it on the asterisk machine. So I don't know if "sl_send_reply" works. Thanks Thorsten
samuel wrote:
Most probably your SER instance does not modify the callerid info so I would check both your asterisk configs and the configuration of your UAs. The ringback tone also looks like a configuration issue of your asterisk.
I would recommend you to get some info about the asterisk configuration to know which the problem might be.
Sam.
2008/4/29, Thorsten <serusers@thorko.de mailto:serusers@thorko.de>:
Hi guys, I'm trying to set up a SER server between 2 asterisk machines. I run into 2 issues. Whenever I call someone I don't get any ringback tone even so the
call
initiating asterisk machine gets the 180 message after 100. <--- SIP read from 10.4.1.80:5060 <http://10.4.1.80:5060> ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134 <mailto:sip%3A1000@82.98.89.134>>;tag=as4c964973 To: <sip:017683035400@10.4.1.80
mailto:sip%3A017683035400@10.4.1.80>
Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 <mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134> CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 <http://10.4.1.80:5060> "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 <http://82.98.89.134> req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80> out_uri=sip:017683035400@192.168.13.102:5060 <http://sip:017683035400@192.168.13.102:5060> via_cnt==1" <-------------> --- (9 headers 0 lines) --- mg03*CLI> <--- SIP read from 10.4.1.80:5060 <http://10.4.1.80:5060> ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060 From: "Thorsten" <sip:1000@82.98.89.134 <mailto:sip%3A1000@82.98.89.134>>;tag=as4c964973 To: <sip:017683035400@10.4.1.80
mailto:sip%3A017683035400@10.4.1.80>;tag=59cea6e4c6ca71e2f82c9c3c8b464af6. bec2
Call-ID: 5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134 <mailto:5e209fbb7ebdbad97f0193515c5a2982@82.98.89.134> CSeq: 102 INVITE Server: Sip EXpress router (0.9.7 (i386/linux)) Content-Length: 0 Warning: 392 10.4.1.80:5060 <http://10.4.1.80:5060> "Noisy feedback tells: pid=459 req_src_ip=82.98.89.134 <http://82.98.89.134> req_src_port=5060 in_uri=sip:017683035400@10.4.1.80 <mailto:sip%3A017683035400@10.4.1.80> out_uri=sip:017683035400@192.168.13.102:5060 <http://sip:017683035400@192.168.13.102:5060> via_cnt==1 On SER I've configured to send this message: if (method=="INVITE") { if (uri =~ "sip:0[0-9]@*") { route(3); sl_send_reply("180", "Ringing"); break; } }; The other issue is that I don't see the caller id on the receiver side. I don't know if it is a asterisk or a SER issue. Only if I set the caller id on asterisk manual in extensions.conf with exten => _X.,1,Set(CALLERID(num)=06965006100) I'll see the caller id on the receiver side. I would really appreciate any help Thanks Thorsten _______________________________________________ Serusers mailing list Serusers@lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
Alberto Cruz -
samuel -
Thorsten