Hi,
Here is a very interesting project which allows ser to run on Linksys wrt54g.
http://sipath.sourceforge.net/
Richard
Why would somebody setup ser on linksys? Anybody know what are the applications?
Thanks
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. IT Manager CCNP, MCSE Security "Secure your self by securing your System" CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115
-----Original Message----- From: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Sent: Wednesday, January 12, 2005 3:04 AM To: serusers@lists.iptel.org Subject: [Serusers] ser on linksys router
Hi,
Here is a very interesting project which allows ser to run on Linksys wrt54g.
http://sipath.sourceforge.net/
Richard
_______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
I can think of two application which might be appealing.
The first one is a pbx which can be deployed in a company. All internal calls are routed through it. One can distribute the central ser server functions into multiple smaller ser servers.
The second one is for nat. For internal calls, there is no nat issue. For external calls, the router can be sit on both inside and outside on the network. You can avoid the nat issue on SIP messages.
Richard
-----Original Message----- From: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of CM Rahman Jr. Sent: Wednesday, January 12, 2005 8:40 AM To: serusers@lists.iptel.org Subject: RE: [Serusers] ser on linksys router
Why would somebody setup ser on linksys? Anybody know what are the applications?
Thanks
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. IT Manager CCNP, MCSE Security "Secure your self by securing your System" CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115
-----Original Message----- From: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Sent: Wednesday, January 12, 2005 3:04 AM To: serusers@lists.iptel.org Subject: [Serusers] ser on linksys router
Hi,
Here is a very interesting project which allows ser to run on Linksys wrt54g.
http://sipath.sourceforge.net/
Richard
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
Richard wrote:
I can think of two application which might be appealing.
The first one is a pbx which can be deployed in a company. All internal calls are routed through it. One can distribute the central ser server functions into multiple smaller ser servers.
I guess for PBX applications asterisk is better.
The second one is for nat. For internal calls, there is no nat issue. For external calls, the router can be sit on both inside and outside on the network. You can avoid the nat issue on SIP messages.
NAT issues can be solved using the ip_conntrack_sip kernel module which is included in OpenWRT (it's not compiled by default, but a friend of mine built a new kernel and it works fine (no STUN or rtpproxy needed any more)
regards, klaus
Richard
-----Original Message----- From: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of CM Rahman Jr. Sent: Wednesday, January 12, 2005 8:40 AM To: serusers@lists.iptel.org Subject: RE: [Serusers] ser on linksys router
Why would somebody setup ser on linksys? Anybody know what are the applications?
Thanks
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&* C.M. Rahman Jr. IT Manager CCNP, MCSE Security "Secure your self by securing your System" CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115
-----Original Message----- From: serusers-bounces@iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Sent: Wednesday, January 12, 2005 3:04 AM To: serusers@lists.iptel.org Subject: [Serusers] ser on linksys router
Hi,
Here is a very interesting project which allows ser to run on Linksys wrt54g.
http://sipath.sourceforge.net/
Richard
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
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
The second one is for nat. For internal calls, there is no nat issue.
For
external calls, the router can be sit on both inside and outside on the network. You can avoid the nat issue on SIP messages.
NAT issues can be solved using the ip_conntrack_sip kernel module which is included in OpenWRT (it's not compiled by default, but a friend of mine built a new kernel and it works fine (no STUN or rtpproxy needed any more)
This sip ALG has some problems. It is work ok with one phone behind NAT, but not multiple phones. SIP is very complicated; probably part of reason why there is no SIP conntrack module available in POM yet.
Richard
Richard wrote:
The second one is for nat. For internal calls, there is no nat issue.
For
external calls, the router can be sit on both inside and outside on the network. You can avoid the nat issue on SIP messages.
NAT issues can be solved using the ip_conntrack_sip kernel module which is included in OpenWRT (it's not compiled by default, but a friend of mine built a new kernel and it works fine (no STUN or rtpproxy needed any more)
This sip ALG has some problems. It is work ok with one phone behind NAT, but not multiple phones. SIP is very complicated; probably part of reason why there is no SIP conntrack module available in POM yet.
Then maybe it's better to use user-mode solutions like http://siproxd.sourceforge.net/
klaus
I can think of two application which might be appealing.
The first one is a pbx which can be deployed in a company. All internal calls are routed through it. One can distribute the central ser server functions into multiple smaller ser servers.
I guess for PBX applications asterisk is better.
In my experience, * is not something I'd recommend to just anyone. Unless one is willing to spend significant amount of time to enhance the code...
Richard
Richard wrote:
I can think of two application which might be appealing.
The first one is a pbx which can be deployed in a company. All internal calls are routed through it. One can distribute the central ser server functions into multiple smaller ser servers.
I guess for PBX applications asterisk is better.
In my experience, * is not something I'd recommend to just anyone. Unless one is willing to spend significant amount of time to enhance the code...
Richard
That's true - but for simple "basic call" applications and it works. And it is the only solution if I have a SIP phone which only supports one account and I want to be registered to severel proxies and want to use them (LCR, eg. nikotel, sipgate ...)
klaus
Hi to all can I change the serctl file and make it use my Postgres database or it just supports MySQL ?? Thanks
João Pereira
Hi
I am using 2 xlite sessions of different PC's, and also a IP phone, now the xlites when they register have the correct IP address in their contact headers, basically they have the FQDN for my sip domain, however the IP phone, it seems to be picking something else up.
The IP address in Call-ID for the xlites is the domain name, but in the IP phone its a IP address of the phone itself, no probs there, but the IP address shwn in the Contact header is 10.44.100.1...have no clue where this is coming from, or where to force it to change...any ideas
Iqbal
Hi Klaus,
On Thursday 13 January 2005 22:38, Klaus Darilion wrote:
Richard wrote:
I can think of two application which might be appealing.
The first one is a pbx which can be deployed in a company. All internal calls are routed through it. One can distribute the central ser server functions into multiple smaller ser servers.
I guess for PBX applications asterisk is better.
That's interesting. I tried to build up my private home pbx with a combination of SER and ASTERISK. The result: The missing call routing capabilities which * describes with the nice word "hairpin" makes it useless for PBX architectures. ASTERISK might be useful for something especially because it supports NT mode for ISDN, but I cannot see for what. Maybe you can help me a little and tell me how to overcome this big "hairpin" issue. I would be very thankful because that would allow me to set up a very nice PBX replacement which was almost completed before I was stopped by the "hairpin".
Thank you, Alex
Alexander Hoffmann wrote:
Hi Klaus,
On Thursday 13 January 2005 22:38, Klaus Darilion wrote:
Richard wrote:
I can think of two application which might be appealing.
The first one is a pbx which can be deployed in a company. All internal calls are routed through it. One can distribute the central ser server functions into multiple smaller ser servers.
I guess for PBX applications asterisk is better.
That's interesting. I tried to build up my private home pbx with a combination of SER and ASTERISK. The result: The missing call routing capabilities which
- describes with the nice word "hairpin" makes it useless for PBX
architectures. ASTERISK might be useful for something especially because it supports NT mode for ISDN, but I cannot see for what. Maybe you can help me a little and tell me how to overcome this big "hairpin" issue. I would be very thankful because that would allow me to set up a very nice PBX replacement which was almost completed before I was stopped by the "hairpin".
What do you mean with hairpin? AFAIK Cisco uses this term and means "VoIP<->VoIP" calls in their gateways. Hairpin is also used in STUN terminology and means that the NAT router forwards packets from inside1 to inside2 although the packets are addressed to the external socket of inside2.
What does hairpin in asterisk terminology mean?
regards, klaus
Hello Klaus,
On Friday 14 January 2005 12:53, Klaus Darilion wrote:
Alexander Hoffmann wrote:
Hi Klaus,
On Thursday 13 January 2005 22:38, Klaus Darilion wrote:
Richard wrote:
I can think of two application which might be appealing.
The first one is a pbx which can be deployed in a company. All internal calls are routed through it. One can distribute the central ser server functions into multiple smaller ser servers.
I guess for PBX applications asterisk is better.
That's interesting. I tried to build up my private home pbx with a combination of SER and ASTERISK. The result: The missing call routing capabilities which * describes with the nice word "hairpin" makes it useless for PBX architectures. ASTERISK might be useful for something especially because it supports NT mode for ISDN, but I cannot see for what. Maybe you can help me a little and tell me how to overcome this big "hairpin" issue. I would be very thankful because that would allow me to set up a very nice PBX replacement which was almost completed before I was stopped by the "hairpin".
What do you mean with hairpin? AFAIK Cisco uses this term and means "VoIP<->VoIP" calls in their gateways. Hairpin is also used in STUN terminology and means that the NAT router forwards packets from inside1 to inside2 although the packets are addressed to the external socket of inside2.
What does hairpin in asterisk terminology mean?
This is my situation: I use one ISDN card in NT mode and connect an ISDN-analog converter to it, in order to use analog phones. The card is controlled by Asterisk and it is configured to route any call established by an analog phone to the SER on the same machine. There is another ISDN card in this computer used to receive incoming PSTN calls. This card is also controlled by asterisk and it is also configured to route all incoming calls to SER. This setup works very well but there is one big issue: If you pick up the analog phone, the call goes through * to SER. Now if you want to place a PSTN call, SER forwards to * in order to reach PSTN. Asterisk then reports a loop which is not correct ! The reason why this does not have to be a loop is that the fact that an INV for the same call hits * twice does not necessarily mean to have a loop. In our case also the direction is important: * is configured to route all INV from the ISDN cards to SER but INVs coming from SER are terminated at the ISDN cards. Thus it is absolutely ok if there is an INV from ISDN card1 -> going to SER -> returning to * -> terminated at ISDN card2. You can easily configure * to route calls like this, but it will complain about loops. If you comment out the loop detection in the source code (because this is not a loop here) then Asterisk will run into a deadlock. I googled for a solution and saw people discussing similar issues and talking about hairpins. IMHO: What ever you call this, it stays a bug in ASTERISK. If you have any suggestions what to do, please let me know !
Thanks, Alex
Hi Alexander!
Alexander Hoffmann wrote:
Hello Klaus,
On Friday 14 January 2005 12:53, Klaus Darilion wrote:
Alexander Hoffmann wrote:
Hi Klaus,
On Thursday 13 January 2005 22:38, Klaus Darilion wrote:
Richard wrote:
I can think of two application which might be appealing.
The first one is a pbx which can be deployed in a company. All internal calls are routed through it. One can distribute the central ser server functions into multiple smaller ser servers.
I guess for PBX applications asterisk is better.
That's interesting. I tried to build up my private home pbx with a combination of SER and ASTERISK. The result: The missing call routing capabilities which * describes with the nice word "hairpin" makes it useless for PBX architectures. ASTERISK might be useful for something especially because it supports NT mode for ISDN, but I cannot see for what. Maybe you can help me a little and tell me how to overcome this big "hairpin" issue. I would be very thankful because that would allow me to set up a very nice PBX replacement which was almost completed before I was stopped by the "hairpin".
What do you mean with hairpin? AFAIK Cisco uses this term and means "VoIP<->VoIP" calls in their gateways. Hairpin is also used in STUN terminology and means that the NAT router forwards packets from inside1 to inside2 although the packets are addressed to the external socket of inside2.
What does hairpin in asterisk terminology mean?
This is my situation: I use one ISDN card in NT mode and connect an ISDN-analog converter to it, in order to use analog phones. The card is controlled by Asterisk and it is configured to route any call established by an analog phone to the SER on the same machine. There is another ISDN card in this computer used to receive incoming PSTN calls. This card is also controlled by asterisk and it is also configured to route all incoming calls to SER. This setup works very well but there is one big issue: If you pick up the analog phone, the call goes through * to SER. Now if you want to place a PSTN call, SER forwards to * in order to reach PSTN. Asterisk then reports a loop which is not correct ! The reason why this does not have to be a loop is that the fact that an INV for the same call hits * twice does not necessarily mean to have a loop. In our case also the direction is important: * is configured to route all INV from the ISDN cards to SER but INVs coming from SER are terminated at the ISDN cards. Thus it is absolutely ok if there is an INV from ISDN card1 -> going to SER -> returning to * -> terminated at ISDN card2. You can easily configure * to route calls like this, but it will complain about loops. If you comment out the loop detection in the source code (because this is not a loop here) then Asterisk will run into a deadlock. I googled for a solution and saw people discussing similar issues and talking about hairpins. IMHO: What ever you call this, it stays a bug in ASTERISK. If you have any suggestions what to do, please let me know !
Yes, I'm aware of this problem - this is IMO a bug in asterisk. This scenarios is not a "loop", but a "spiral" and spirals should be allowed. Its probably because af the bad dialog matching code in asterisk. I would suggest to use an ATA (SIPURA, Grandstream) instead of asterisk to connect an anlog phone.
regards, klaus
Hello,
--- Alexander Hoffmann alexander.hoffmann@netgenius.de wrote:
<SNIP>
This setup works very well but there is one big issue: If you pick up the analog phone, the call goes through * to SER. Now if you want to place a PSTN call, SER forwards to * in order to reach PSTN. Asterisk then reports a loop which is not correct ! The reason why this does not have to be a loop is that the fact that an INV for the same call hits * twice does not necessarily mean to have a loop. In our case also the direction is important: * is configured to route all INV from the ISDN cards to SER but INVs coming from SER are terminated at the ISDN cards. Thus it is absolutely ok if there is an INV from ISDN card1 -> going to SER -> returning to * -> terminated at ISDN card2.
Instead of forwarding calls back to Asterisk from SER, reply Asterisk with a '302 Moved Temporarily' with a modified contact header. I think it should solve your problem.
Regards,
===== Girish Gopinath gr_sh2003@yahoo.com
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-
Alexander Hoffmann -
CM Rahman Jr. -
Girish -
Iqbal -
Joao Pereira -
Klaus Darilion -
Richard