c - Error: pjsip gives `mod_sofia` insetad of the actual phone number

Question

I am working on project in which I use Sipek Voip for connecting to Freeswitch. Here is the situation:

I have a Sangoma A400 hard. I compiled Freeswitch for Windows and now it works perfectly.
I have also created a Softphone using Sipek Voip SDK and it works well with Freeswitch.

The problem is that, when I have an incoming call, instead of showing the callers number, I get mod_sofia.

I looked at Sipek and all it gets from pjsip is a string containing <sip:mod_sofia@192.168.2.10:5060>.

So I went to pjsip and tried to pass the actual phone number to Sipek. I found out there is a function called pjsua_call_on_incoming which handles an incoming call.

It takes an argument of type pjsip_rx_data. It has a string field (rdata->msg_info.msg_buf) which contains the whole message. I tried to replace <sip:mod_sofia@192.168.2.10:5060> with the actual number, but it has no effect.

Does anyone have any idea how to fix this?

Answer 1

您可以查看此链接以跟踪问题。不幸的是，即使在论坛上，也几乎没有人可以“免费”帮助您解决开源项目。我只是从我的个人经历来说。我面临着同样的问题，直到现在都无法弄清楚，尽管我已经解决了我过去在 SIPEK 上遇到的许多问题，这一切都是我自己解决的。

Answer 2

添加sip_contact_user=xxxx您的拨号字符串。

Answer 3

我不明白您问题的根源在于 FreeSWITCH 或 sipek/pjsip。FreeSWITCH wiki 上的此条目可以帮助您调试 FreeSWITCH 中的 sip 堆栈： http ://wiki.freeswitch.org/wiki/Mod_sofia#Debugging_Sofia-SIP ，其方式类似于wireshark 捕获。很抱歉，我不知道如何帮助您追踪 pjsip 中 msg_info.msg_buf 的解析/渲染。